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133,093 Commits 1 Branch 166 Tags 1.2 GiB
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421 Commits

Author SHA1 Message Date
Randell Jesup
eb99b87929 Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM r=derf 2013-01-29 11:55:09 -05:00
Boris Zbarsky
de6a67d7d0 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
Phil Ringnalda
b48b7c8bbd Back out c5238879470f and b3cabb259af7 (bug 830099) for bustage 2013-01-28 22:36:59 -08:00
Boris Zbarsky
daed28912a 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
Ethan Hugg
eeae54b2b6 Bug 835290 MediaPipeline - replace attempted += of ints with PR_snprintfs r=jesup 2013-01-28 12:26:32 -08:00
Randell Jesup
f82f847448 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
EKR
d005034d5d Bug 816780 - Merge all incoming m-lines into one MediaStream. r=jesup,abr 2013-01-24 08:34:18 -08:00
Justin Lebar
da5a042a4d 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
Jan-Ivar Bruaroey
e473c4cd28 Bug 834463: Corrected RTCConfiguration format. r=jst 2013-01-24 17:58:29 -05:00
Jan-Ivar Bruaroey
327a61e43e Bug 825703: Stun configuration from JS for PeerConnections (IP only) r=bz,jesup 2013-01-23 14:21:25 -05:00
EKR
63eedb4bb4 Bug 786236: Per-context configurable STUN servers. r=abr 2013-01-05 11:40:34 -08:00
Randell Jesup
6c2f1e7349 Bug 832683: Match SRTP policy values to enable NACK mode in webrtc r=ekr 2013-01-23 16:41:35 -05:00
Steven Lee
18cbda81f3 Bug 818843 - Media changes. r=rjesup 2013-01-23 18:08:16 -05:00
Ethan Hugg
72d22cb1f8 Bug 806825 - Fix possible data race of PeerConnectionCtx::mSipccState r=jesup 2013-01-22 11:25:04 -08:00
EKR
977c5c9e29 Bug 829757 - Clean up a=candidate stripping. r=abr 2013-01-19 15:18:28 -08:00
Ehsan Akhgari
893e6d1c91 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
Ehsan Akhgari
e52df68fa4 Bug 833118 - Disable PGO on webrtc; a=me 2013-01-21 18:02:12 -05:00
Randell Jesup
b41be96585 Bug 832567: Adjust minimum default bitrate to 200Kbps until we support lower resolutions rs=me 2013-01-21 03:49:57 -05:00
Randell Jesup
316409980c 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
Nicholas Nethercote
0293b60b78 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
EKR
1527ae263c Bug 831764. Reverse DTLS client/server selection to more closely match RFC 5763. r=jesup 2013-01-17 07:54:53 -08:00
EKR
d7abaa8a5a Bug 829624 - Always use 80-bit MAC with SRTCP. r=jesup 2013-01-18 13:03:17 -08:00
Randell Jesup
5b31935363 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
Adam Roach [:abr]
9c2697eb77 Bug 824919: Weaken key references to PeerConnection and friends r=jesup,smaug,ekr 2013-01-17 17:11:14 -06:00
EKR
2d62e13f38 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
Adam Roach [:abr]
42bdaeb63d Bug 829461: Check for NULL media stream r=ehugg 2013-01-17 15:32:16 -06:00
Randell Jesup
5e909176f9 Bug 826807: Clean up JSAPI error handling in PeerConnection constraints r=bz 2013-01-17 16:23:44 -05:00
Ethan Hugg
594892fa69 Bug 822197 Handle creation of more PeerConnections than MAX_LINES r=jesup 2013-01-15 13:11:47 -08:00
Ryan VanderMeulen
1569c1b539 Backed out changeset cebb008a72f9 (bug 824919) for mochitest b-c orange. 
CLOSED TREE
 2013-01-14 22:01:12 -05:00
Adam Roach [:abr]
effbd6b1c4 Bug 824919: Weaken key references to PeerConnection and friends,r=jesup,smaug,ekr 2013-01-14 17:00:20 -06:00
Adam Roach [:abr]
6c6d18e34d Bug 829761: Remove inactive check for sdpmode if port != 0, r=ekr 2013-01-14 16:38:38 -06:00
EKR
b6b0bb54c1 Bug 828027 - Lower-case digest algorithm names r=ehugg 2013-01-14 09:10:38 -08:00
Chris Peterson
eecfc92b29 Bug 829344 - Part 3: Fix gcc -Wlogical-op warnings in SIP code. r=jesup 2013-01-10 15:19:26 -08:00
Robert O'Callahan
f2ee4df1a0 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
Chris Peterson
d2b4e410e5 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
Gian-Carlo Pascutto
32c6ea4245 Bug 827359 - Disable unconditional use of NEON code in WebRTC DSP code. r=jesup 2013-01-08 18:30:10 +01:00
Ehsan Akhgari
129c4cd7fd Bug 579517 follow-up: Remove NSPR types that crept in 2013-01-07 18:21:50 -05:00
Doug Turner
514ed315e4 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
Steven Lee
aa0230cfd3 Bug 818843 - Media changes. r=rjesup 2013-01-06 22:24:14 -05:00
Randell Jesup
9c7dcac8d7 Bug 805251 - Minimal fix for creation refcount and transport release r=ekr,derf 2013-01-06 22:01:23 -05:00
Ethan Hugg
9d4b237c16 Bug 825785 Signaling - monitor threads for shutdown r=jesup 2013-01-03 19:12:28 -08:00
EKR
51e907a976 Bug 825611 - Fix unit tests to match fix to MediaPipeline. r=jesup 2013-01-05 14:52:11 -08:00
EKR
ec35a68546 Bug 825611 - Have MediaPipeline deliver as much media as requested. r=jesup 2013-01-05 09:02:36 -08:00
Jan-Ivar Bruaroey
7fc1ad4d05 Bug 820538: Added media-ptr check because of NS_DISPATCH_SYNC. r=rjesup 2013-01-04 23:01:56 -05:00
Maire Reavy
23be056db4 bug 822956 - correct chunk_remaining calculation r=ekr 2013-01-04 14:41:25 -05:00
Robert O'Callahan
83308df973 Bug 822956. ProcessAudioChunk needs to take account of AudioChunk::mOffset. r=jesup 2013-01-04 12:16:32 -05:00
Maire Reavy
fbc9c3a8b2 bug 822956: Handle audio chunks that aren't 10ms in duration r=ekr 2013-01-03 18:37:55 -05:00
EKR
95c13817cf Bug 826529 - Increase default video bitrate. r=jesup 2013-01-03 19:17:42 -08:00
Randell Jesup
cda946d552 Bug 826008: Fix type tests for constraints objects r=smaug 2013-01-03 11:58:34 -05:00
Adam Roach [:abr]
68dc4ef868 Bug 825086: Removing errant free of body parts when encoding fails r=ekr 2013-01-02 15:06:12 -06:00
Ethan Hugg
2a5dbe281d Bug 825785 cprDestroyThread on Windows should kill thread immediately r=jesup 2013-01-02 17:08:48 -08:00
EKR
4c488b7543 Bug 825439: Disconnect pipeline on STS Thread r=jesup 2012-12-29 09:24:34 -08:00
Jan-Ivar Bruaroey
1769b4d97e Bug 824263 - Shutdown: PeerConnectionMedia disconnect_all() + peer_ctx cleanup. r=jesup,ekr 2012-12-27 17:38:45 -05:00
Randell Jesup
80389454ef Bug 825526: Improve lifetime control of SourceMediaStream in gUM r=anant 2012-12-31 18:12:15 -05:00
Adam Roach [:abr]
e1679c7d8f Bug 821071: Initialize all out parameters in VcmSIPCCBinding.cpp, r=ekr 2012-12-31 11:43:22 -06:00
Randell Jesup
873565fa76 Bug 825514: Add safety check to PeerConnectionCtx shutdown r=ekr 2012-12-31 12:34:44 -05:00
Adam Roach [:abr]
f6e86aeb0f Bug 824956: Activate media type when set to receive media by constraint r=jesup 2012-12-28 16:40:51 -06:00
EKR
ff97d84f02 Bug 825336: Add constraint to suppress data channel r=jesup 2012-12-28 12:09:58 -08:00
EKR
0357be55dc Bug 825477 - Ignore return value from async calls in MediaPipeline. r=jesup 2012-12-30 07:53:48 -08:00
Adam Roach [:abr]
ebca40ad6a Bug 825106: Use the correct pointer comparison when determining master r=jesup 2012-12-28 15:12:47 -06:00
Randell Jesup
a95ef36414 Bug 820011: Shut down webrtc signling service on XPCOM shutdown r=derf 2012-12-28 17:04:32 -05:00
Adam Roach [:abr]
a309ab83a5 Bug 824220: Pass constraints through directly through message, deallocate when done r=jesup 2012-12-27 14:28:11 -06:00
Jan Beich
a263805ced Bug 815916 - Unbreak building signaling tests for system jpeg/pixman. r=rjesup 2012-12-26 10:15:53 -05:00
Nicholas Nethercote
4a4db06c69 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
EKR
97cf51bbe1 Bug 824351 - Make PeerConnection.Close not check state. r=ehugg 2012-12-23 17:03:26 -08:00
Adam Roach [:abr]
2f7c8fe8a0 Bug 821003: Replace snprintf macro with static function r=jesup 2012-12-21 14:55:41 -06:00
Adam Roach [:abr]
08683921f1 Bug 818714: Set media enabled to FALSE unless added using addStream, r=ehugg 2012-12-20 16:05:28 -06:00
Adam Roach [:abr]
397a8c9ff4 Bug 797534: Update PeerConnectionImpl to use SDP from SIPCC, r=ehugg 2012-12-19 09:25:51 -06:00
Jan-Ivar Bruaroey
2088668bc4 Bug 822158: Use async dispatch of Ice(Gathering)Completed to unwind stack r=jesup,ekr 2012-12-21 15:21:15 -05:00
Dan Mosedale
b74e965a62 Bug 819856, Re-enabled WebRTC voice engine code for Android, r=jesup 2012-12-21 14:03:33 -08:00
EKR
3f8b6eafea Bug 820102 - Clean up MediaPipeline threading (re-land after fix). r=derf,jesup 2012-12-21 06:03:22 -08:00
Dan Mosedale
e6b46e87fb 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
Ed Morley
eaecff5d2f Backout 21409a401d75 (bug 821292),9587e39f9a50 (bug 820102) for conflicts and assertions respectively, on a CLOSED TREE 2012-12-21 16:15:01 +00:00
EKR
a35760e672 Bug 820102 - Clean up MediaPipeline threading. r=derf,jesup 2012-12-21 06:03:22 -08:00
Henrik Skupin
9b2002d57a Backout bug 818714 because it introduces a sigabort crash 
--HG--
extra : rebase_source : 4307433661215827bc238f514fa66758e27366bc
 2012-12-20 13:47:12 +01:00
EKR
599486b77f Bug 817430: Set initial controlled/controlling values, based on inbound T/F. r=jesup 2012-12-02 13:49:56 -08:00
Adam Roach [:abr]
fd393200d4 Bug 818714: Set media enabled to FALSE unless added using addStream, r=ehugg 2012-12-19 20:52:32 -06:00
Jan-Ivar Bruaroey
352cbc5c20 Bug 794240 followup: Fixes implicit function warning that broke linkage + symmetric fixes r=jesup 2012-12-19 13:00:49 -05:00
Randell Jesup
57f3ac27ab 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
Jan-Ivar Bruaroey
cf73ed4ca2 Bug 794240: Disable timerthread + close sockets + cleanup tmp-files on shutdown + file permissions. r=jesup 2012-12-14 16:15:21 -05:00
Ethan Hugg
faef85a350 Bug 806829 - Signaling - remove g_deviceInfo.name r=jesup 2012-12-14 13:43:06 -08:00
Ted Mielczarek
63ad63bf4b 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
Ethan Hugg
eba9ddac43 Bug 807012 - Signaling - reorder initiialization of CallControlManagerImpl r=jesup 2012-12-17 11:01:32 -08:00
Chris Peterson
28a46a762a Bug 821621 - Fix unused variable warning in PeerConnectionCtx.cpp. r=jesup 2012-12-15 00:22:47 -08:00
Randell Jesup
df518b4d88 Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus 2012-12-13 23:30:11 -05:00
Randell Jesup
456f66acde backout 916f6915112d bug 806375 for anonymous enum bustage on linux 2012-12-14 00:26:58 -05:00
Randell Jesup
07abce4f80 Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus 2012-12-13 23:30:11 -05:00
Gian-Carlo Pascutto
cd89c2e152 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
Gian-Carlo Pascutto
6c7bf909a2 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
Ethan Hugg
d1c6d3a72d Bug 820550 - Signaling - stop checking API state in CloseInt r=ekr 2012-12-12 09:39:51 -08:00
EKR
7468199935 Bug 820671 Make PCImpl a tombstone if mMedia is null r=jesup 2012-12-11 19:45:09 -08:00
Dan Mosedale
b0b771805a 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
Ted Mielczarek
201365152b bug 820769 - fix mozmake.py to generate correct common.mk include. r=jesup 2012-12-12 11:32:05 -05:00
EKR
668853d1d7 Bug 792175 - Move PeerConnection operations onto main thread. r=jesup,derf,ehugg 2012-09-26 10:14:23 -07:00
Gian-Carlo Pascutto
6b2ab16f81 Bug 816575 - Remove call to SetRenderAndroidVM when not using internal renderer. r=tterribe 2012-12-11 20:11:15 +01:00
Gian-Carlo Pascutto
47d604cb18 Bug 813913 - WebRTC signaling code doesn't yet build on Android. r=dmose rs=dmose 2012-12-10 16:12:50 -08:00
Randell Jesup
d8a72757fb Bug 752657: backout sigslot multithreaded due to deadlocks rs=jesup 2012-12-11 12:54:24 -05:00
Randell Jesup
741c9bfc87 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
Randell Jesup
8ac21346c0 Bug 806822: avoid race condition in gathering stats r=ehugg 2012-12-11 07:58:53 -05:00
Steven Lee
97a8e887c0 Bug 819246 - arm_neon is meaningful only when armv7 is true. r=jesup 2012-12-10 09:05:00 -05:00
Ed Morley
d5e8c2c0a2 Merge mozilla-central to mozilla-inbound 2012-12-10 14:03:11 +00:00
Mike Hommey
898e36d822 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
EKR
bc440a4092 Bug 816112: Configure webrtc video/RTCP parameters for better defaults r=jesup 2012-11-28 08:34:10 -08:00
Oleg Romashin
09cb51df19 Bug 819364 - Webrtc signaling unit test does not build on Qt port. r=rjesup 2012-12-09 09:28:37 -08:00
Saurabh Anand
151d13c7a5 Bug 818817 - Fix some compiler warnings, r=Ms2ger 2012-12-09 22:53:19 +05:30
Adam Roach [:abr]
b71e4a0fb9 Bug 817065: Replace vcm_media_payload_t with structure leveraging rtp_ptype constants r=ehugg,jesup 2012-12-06 11:36:43 -06:00
Suhas Nandakumar
23b591f2df Bug 817488 Audio Send and Recv support for PC tests r=ekr 2012-12-05 13:37:32 -08:00
Mike Hommey
4125bfa14e Bug 814693 - Allow webrtc to build on more architectures. r=jesup 
--HG--
extra : rebase_source : 7d22643c1b4b944595bfe33ac8f5925af2b8bcbd
 2012-12-01 09:55:48 +01:00
Ethan Hugg
6b5299e68b Bug 814329 Protect fim_process_event from NULL input r=jesup 2012-11-28 08:38:50 -08:00
Gian-Carlo Pascutto
27a071628d Bug 750869 - Build system support for --enable-webrtc for Android (off by default). r=ted 2012-12-04 16:27:18 +01:00
Makoto Kato
cb4a57156c Bug 817481 - Build failure on WebRTC unit test with --with-system-libvpx. r=jesup 2012-12-03 17:01:16 +09:00
Crypt
77bd1ae21d Bug 810220 - Patch to fix SDP Codec Negotiation Issues (revised) r=ekr,jesup 2012-11-26 02:36:43 -08:00
EKR
8243b8715e Bug 817451, Correctly set mRole in PC unit tests. r=ehugg 2012-12-02 18:06:21 -08:00
Brian Smith
be0d41bee2 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
Randell Jesup
6a6982acf8 Bug 812636: additional checks for failure for a Conduit be created r=ehugg 2012-11-28 03:28:30 -05:00
Daniel Holbert
e548ef3a86 Bug 815928 part 2: Fix typo s/elment/elment/ in comments & tests. DONTBUILD, rs=Waldo 2012-11-27 19:15:36 -08:00
Daniel Holbert
6044359950 Bug 815928 part 1: Fix typo s/elelment/element/ in comments & tests. DONTBUILD, rs=Waldo 2012-11-27 19:15:35 -08:00
Suhas Nandakumar
84e358a764 Bug 814734 - Fixed Log format string mismatches r=ekr 2012-11-26 08:38:14 -08:00
EKR
9f3752906a Bug 805662, Crash in mediaconduit_unittest. r=jesup 2012-11-22 10:57:37 -08:00
Adam Roach
7aae524c33 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
Boris Zbarsky
59ee5f728d Bug 813746. Link the webrtc unit tests to ZLIB as needed. r=bz 2012-11-21 02:16:03 -05:00
Ethan Hugg
24523e6913 Bug 813723 Reorder state assignment in unit tests r=ekr 2012-11-20 13:47:07 -08:00
Ethan Hugg
8a9912197c Bug 811118 build webrtc unittests by default but run only some r=jesup 2012-11-12 15:34:05 -08:00
Ethan Hugg
66c75c4055 Bug 813212 Update signaling unittests to match latest IDL r=jesup 2012-11-19 09:59:36 -08:00
EKR
4e3192c90d Bug 799419: Force NSS startup during PeerConnection Initialize r=bsmith,jesup 2012-11-18 07:43:26 -08:00
Randell Jesup
8c4148d730 Bug 812886: Watch network (tear)down events and kill PeerConnections r=ekr,smaug,bsmith 2012-11-18 23:53:14 -05:00
Randell Jesup
832b4cf1ef Bug 812641: Shut down SipCC instance when number of PeerConnections == 0 (reland) r=ehugg 2012-11-17 23:03:58 -05:00
Randell Jesup
932db553b0 Bug 812641: Shut down SipCC instance when number of PeerConnections==0 r=ehugg 2012-11-18 01:42:40 -05:00
Ryan VanderMeulen
ba8b11631a Backed out changeset b527670b6728 (bug 811118) for orange. 2012-11-16 21:24:14 -05:00
Nathan Froyd
9213a43bcf Bug 810544 - don't include <iostream> in webrtc code where it's not necessary; r=jesup 2012-11-09 11:08:22 -05:00
Ethan Hugg
a630f04b9d Bug 811118 build webrtc unittests by default r=jesup 2012-11-12 15:34:05 -08:00
Randell Jesup
7e9bfa7919 Bug 806830: Enforce initializing strlib before using r=ehugg 2012-11-16 14:37:08 -05:00
EKR
b7ac4a3223 Bug 811183: Recursive GC In PeerConnection shutdown; r=jesup 2012-11-16 10:27:30 -08:00
Timothy B. Terriberry
1424d481c0 Bug 810363 - Reject non-stereo, non-48 kHz Opus streams, r=ehugg 2012-11-15 15:09:39 -08:00
Timothy B. Terriberry
8559fbd229 Bug 810353 - Offer Opus as stereo instead of mono, r=ehugg 2012-11-15 15:09:39 -08:00
Randell Jesup
f01b2fce9c Bug 811695: disable internal socket transports for getUserMedia Audio capture r=derf 2012-11-15 17:58:40 -05:00
Adam Roach
c958d091b9 Bug 803318: Improved handling of constraints and more tests. r=ekr 2012-11-14 11:25:14 -06:00
Ethan Hugg
7617be8bac Bug 811333 Signaling - Fix size of thread ID for Windows build r=jesup 2012-11-13 09:23:04 -08:00
Ethan Hugg
babd8ac35f Bug 806407 Fix type in struct passed into msgsnd f=jesup 2012-11-09 08:47:00 -08:00
Nathan Froyd
426bb1a317 Bug 809950 - fix webrtc signaling Wrapper code to not introduce static initializers; r=jesup 2012-11-08 13:45:39 -05:00
Randell Jesup
825968012c Bug 807929: Make DataChannel refcounted r=mcmanus 2012-11-02 15:28:13 -04:00
EKR
5afde87d6d Bug 807259: Fix shutdown for WebRTC standalone unit tests. r=anant 2012-10-31 16:31:47 +01:00
Randell Jesup
843f3475a9 Bug 805279: make sure we pass values for formatted log messages r=derf 2012-10-30 15:31:09 -04:00
EKR
aa12461aed Bug 806335: MediaPipeline destroys TransportFlow on wrong thread; r=anant 2012-10-29 15:35:12 +01:00
EKR
8a817de1e6 Bug 806306: Fix compile bustage for MediaConduit; r=anant 2012-10-29 15:33:19 +01:00
Robert O'Callahan
a738cb5616 Bug 805703. Part 1: MediaStreamGraph::CreateInputStream -> CreateSourceStream. r=jesup 
--HG--
extra : rebase_source : 3c327e9ead92508f12df4b95f2fd24fa2ba97ab5
 2012-10-29 17:36:31 +13:00
Randell Jesup
0db8fa4187 Bug 805701: protect against NULL fcb pointer r=ehugg 2012-10-27 21:47:39 -04:00
EKR
a0982d385e Bug 801221: Move mtransport operations onto STS thread; r=jesup 2012-10-18 13:01:52 -07:00
Ethan Hugg
8c9b130043 Bug 805533 Send SDP parse errors through single function for reporting r=jesup 2012-10-26 13:05:50 -07:00
Robert O'Callahan
384eb67455 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
Robert O'Callahan
a0d739c463 Bug 805254. Part 7: Move SampleFormat to mozilla::AudioSampleFormat in its own file. r=kinetik 2012-10-25 23:09:40 +13:00
Robert O'Callahan
11cca22e78 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
Ryan VanderMeulen
4c180e35d1 Merge the last PGO-green inbound chnageset to m-c. 2012-10-25 21:14:50 -04:00
Randell Jesup
ea392f39f4 Bug 803842: Fix incorrect index for warning length; bullet-proof the code some r=derf 2012-10-24 15:39:55 -04:00
Ethan Hugg
6d142981c7 Bug 803744 - Allow signaling startup when already started r=jesup 2012-10-24 15:46:40 -07:00
Anant Narayanan
a3a1f3a567 Bug 802694: Pass along constraints from PC JS module to PCImpl; r=ekr,jesup 2012-10-25 12:24:30 -07:00
Ehsan Akhgari
78b2656cbc Backed out 2 changesets (bug 579517) 
Backed out changeset 5298adc70963
Backed out changeset 86ccf7c918ce (bug 579517)
 2012-10-25 12:32:24 -04:00
Ehsan Akhgari
03be701616 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
Randell Jesup
e20b9edb57 Bug 803881: Fix wrong number of samples for webrtc media unit tests r=ekr 2012-10-24 16:54:33 -04:00
Ethan Hugg
af6af2772f Bug 729541 fix for syntax error in signaling_unittests r=jesup 2012-10-23 21:26:27 -07:00
Ethan Hugg
0cb9653ca1 Bug 797512 Signaling: simplify set_dtls_fingerprint r=jesup 2012-10-22 16:38:52 -07:00
Ethan Hugg
04d9d55b98 Bug 803875 MediaConstraints - fix length handling r=jesup 2012-10-22 13:25:44 -07:00
Enda Mannion
8dbb7f02c4 Bug 729541: Remove 'offer' parameter from RTCPeerConnection.createAnswer r=jesup 2012-10-15 16:50:10 +01:00
Randell Jesup
80fe7b4291 Bug 803086: Process NULL image chunks in NotifyQueuedTrackChanges() r=ekr 2012-10-19 10:58:06 -04:00
Enda Mannion
aa298dcb7a Bug 800688: remove local_dynamic_payload_type_value from negotiation r=ehugg,jesup 2012-10-17 15:57:52 +01:00
Ethan Hugg
be98e21abb Bug 798873 Patch 3 - flex_string fix for Windows vsnprintf r=jesup 2012-10-17 18:57:57 -07:00
Mike Hommey
fe09dd0362 Bug 799975 - Fix webrtc when building with system nspr/nss. r=ted 2012-10-17 16:33:23 +02:00
Mike Hommey
c94a36bf30 Bug 798926 - Define INTEGER_TYPES_H and others for srtp.h. r=jesup 2012-10-17 16:28:33 +02:00
Enda Mannion
5c4399b243 Bug 786152: Increment DataChannel SCTP port in a WebRTC JSEP answer r=jesup 2012-09-28 11:36:02 +01:00
Enda Mannion
3e1e925af0 Bug 784515: add hints to webrtc/signaling CreateOffer and CreateAnswer APIs r=jesup 2012-09-28 11:09:09 +01:00
Peter Van der Beken
59b6711296 Fix for bug 711628 (Implement PeerConnection.localStreams/remoteStreams). r=bz. 
--HG--
extra : rebase_source : e21a429d85fc60a972752a2a7deb88a7cb648f3c
 2012-09-13 18:04:31 +02:00
Randell Jesup
df2270282f Bug 800847: Correctly regenerate Makefiles from gyp files, and handle fancy symlinked objdirs r=ted 2012-10-12 16:06:33 -04:00
Ethan Hugg
ec8962d879 Bug 800611 - Trailing whitepace removed from signaling code r=jesup 2012-10-12 08:15:24 -07:00
Gervase Markham
9094e2befd Bug 796457 - upgrade license to MPL 2. 2012-10-12 15:58:11 +01:00
Ed Morley
f63fc6d67a 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
Peter Van der Beken
772951ca5e Fix for bug 711628 (Implement PeerConnection.localStreams/remoteStreams). r=bz. 
--HG--
extra : rebase_source : d7be35954ba69563ed26b2173c75fe9e74707847
 2012-09-13 18:04:31 +02:00
Aryeh Gregor
08caf6e336 Bug 800343 - Cast nsresult to uint32_t to output to streams; r=ehsan 2012-10-11 14:48:31 +02:00
EKR
aef590192f Bug 792811 - Check creation of WebRTC transportflows r=jesup 2012-10-09 12:29:01 -07:00
Ethan Hugg
387cd77603 Bug 800502 - Signaling - Removed unused code that required XML handling r=jesup 2012-10-11 14:07:50 -07:00
Ethan Hugg
c1a2f6f5f9 Bug 800558 - Signaling - return proper error code in fsmdef.c r=jesup 2012-10-11 14:32:45 -07:00
Ethan Hugg
6aedcb664a Bug 798873 - Signaling SDP construction uses flex_string r=jesup 2012-10-08 16:43:33 -07:00
Gregory Szorc
25e7bfc0ac 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
Ehsan Akhgari
d484c73230 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
Ed Morley
3075e97edb Backout d6e84a64d3d4 (bug 798969) for burning 2012-10-11 14:37:44 +01:00
Robert O'Callahan
6f09587042 Bug 798969. Handle symlinked srcdirs by avoiding use of Python's __file__. r=jesup 2012-10-08 17:45:10 +13:00
Ethan Hugg
df8f34e78e Bug 791278: Protect PeerConnection setLocal/RemoteDescription from NULL input r=jesup 2012-10-09 15:14:51 -07:00
Ethan Hugg
17992eed74 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
Randell Jesup
c493d4068a Bug 799477: missing stdarg.h in CSFLog.h (typically gcc 4.4.x) r=ehugg 2012-10-09 14:46:34 -04:00
EKR
57e12aaa01 Bug 799246: Conditionally enable webrtc unit tests r=jesup,ted,cjones 2012-10-08 18:56:00 -07:00
Randell Jesup
35fa2fa89e Bug 799071: clean up clang warnings in media/webrtc/signaling (-Werror) r=ehugg 2012-10-08 17:11:10 -04:00
Ethan Hugg
467783b424 Bug 798873 - Signaling - increase max SDP size (temporary until better fix). r=jesup,ekr 2012-10-07 22:28:52 -07:00
Randell Jesup
af2097d99b BUg 799076: Add temporary error definitions to media/webrtc/signaling for VC9 and below r=derf 2012-10-08 11:08:52 -04:00
Randell Jesup
678bc3bb58 Bug 798892: Wrong enum type in gCCApp.state, breaks bleeding-edge clang builds r=ehugg 2012-10-07 21:55:45 -04:00
Ehsan Akhgari
3239e8a0d5 Bug 579517 follow-up: Remove NSPR types that crept in 2012-10-07 18:26:08 -04:00
Anant Narayanan
ba6106d6fa Bug 798825: Add DataChannel DOM interfaces to RTCPeerConnection; r=smaug 2012-10-07 01:34:30 -04:00
Anant Narayanan
62ba9bbbc0 Bug 694807: Implement PeerConnection C++ module; r=jst,jesup,ekr 2012-10-07 01:34:30 -04:00
EKR
3ba764ed03 Bug 790517: mtransport - Generic media transport subsystem for ICE and DTLS r=jesup,bsmith,mcmanus 2012-10-02 13:04:58 -07:00
Randell Jesup
4550c9cb73 Bug 792188: Fixes to signaling MediaPipeline and Srtpflow r=jesup,derf 2012-10-07 01:34:29 -04:00
Randell Jesup
ad6bad1bb0 Bug 792188: MediaPipeline changes for YUV buffer handling, put ImageContainer back to normal r=ekr 2012-10-04 05:35:26 -04:00
Randell Jesup
5d0a8cd0f2 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
EKR
f6cc1edaee Bug 792188: Import SIPCC into media/webrtc/signaling rs=jesup r=gerv 2012-03-13 11:22:14 -07:00
Randell Jesup
a858a862db 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
Randell Jesup
e271f577fd 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
rename : media/webrtc/trunk/src/voice_engine/main/source/Android.mk => media/webrtc/trunk/src/voice_engine/Android.mk
rename : media/webrtc/trunk/src/voice_engine/main/source/channel.cc => media/webrtc/trunk/src/voice_engine/channel.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/channel.h => media/webrtc/trunk/src/voice_engine/channel.h
rename : media/webrtc/trunk/src/voice_engine/main/source/channel_manager.cc => media/webrtc/trunk/src/voice_engine/channel_manager.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/channel_manager.h => media/webrtc/trunk/src/voice_engine/channel_manager.h
rename : media/webrtc/trunk/src/voice_engine/main/source/channel_manager_base.cc => media/webrtc/trunk/src/voice_engine/channel_manager_base.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/channel_manager_base.h => media/webrtc/trunk/src/voice_engine/channel_manager_base.h
rename : media/webrtc/trunk/src/voice_engine/main/source/channel_unittest.cc => media/webrtc/trunk/src/voice_engine/channel_unittest.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/dtmf_inband.cc => media/webrtc/trunk/src/voice_engine/dtmf_inband.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/dtmf_inband.h => media/webrtc/trunk/src/voice_engine/dtmf_inband.h
rename : media/webrtc/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc => media/webrtc/trunk/src/voice_engine/dtmf_inband_queue.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/dtmf_inband_queue.h => media/webrtc/trunk/src/voice_engine/dtmf_inband_queue.h
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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
foudfou
62d7406c52 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
Chris Jones
d286b9b947 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
Ms2ger
70856e2961 Bug 792343 - Enable FAIL_ON_WARNINGS in some more of dom/ (second batch); r=mounir 2012-09-20 09:55:36 +02:00
Mike Hommey
c1b35e7b2a Bug 774032 bonus - Use @DEPTH@ and @relativesrcdir@ in Makefile.in. r=ted 2012-08-04 20:26:44 +02:00
Andrew Benton
c575567a81 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
Randell Jesup
5588af7472 Bug 694817: Turn off webrtc's internal video renderer and protobuf (take 2) r=ted 2012-07-23 22:35:35 -04:00
Tim Abraldes
1d1133e080 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
Randell Jesup
390730bf91 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
Randell Jesup
7401f0cd15 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
Randell Jesup
4797f62dc3 Bug 770230: Remove broken symlinks in webrtc to third_party/google-visualization-python DONTBUILD r=derf 2012-07-06 12:15:49 -04:00
Randell Jesup
1a9b82b934 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
Randell Jesup
8d9866664c 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
Randell Jesup
435602cf1a Bug 757637: Rollup makesystem changes for webrtc r=khuey r=ted f=glandium 2012-06-20 07:27:43 -04:00
Randell Jesup
8391238a46 Bug 766253: Fix type for kARGBToV table (upstream issue 188 at webrtc.org) r=derf 2012-06-20 07:27:32 -04:00
Randell Jesup
c829262c70 Bug 757637: Rollup media/webrtc/trunk changes from webrtc.org drop r=ted r=derf 2012-06-20 07:27:22 -04:00
Randell Jesup
fa40782d27 bug 731407: need to include <assert.h> in non-debug builds r=cpearce 2012-04-04 14:49:12 -04:00
Bas Schouten
3b2e1d8f19 bug 731407: Remove DShow BaseClass usage from webrtc drop r=cpearce 2012-03-15 23:06:35 +00:00
Randell Jesup
1087aa6fbf 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