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442 Commits

Author SHA1 Message Date
Jan-Ivar Bruaroey
5551c82776 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
a2c6bd8dbd Bug 806829 - Signaling - remove g_deviceInfo.name r=jesup 2012-12-14 13:43:06 -08:00
Ted Mielczarek
93962b8c5e 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
0c7d43972a Bug 807012 - Signaling - reorder initiialization of CallControlManagerImpl r=jesup 2012-12-17 11:01:32 -08:00
Chris Peterson
e7f2e0fc3c Bug 821621 - Fix unused variable warning in PeerConnectionCtx.cpp. r=jesup 2012-12-15 00:22:47 -08:00
Randell Jesup
864277e5f6 Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus 2012-12-13 23:30:11 -05:00
Randell Jesup
a3907b5f58 backout 916f6915112d bug 806375 for anonymous enum bustage on linux 2012-12-14 00:26:58 -05:00
Randell Jesup
c1509f5b67 Bug 806375: cleanup DataChannel, esp. channel close and connection shutdown r=mcmanus 2012-12-13 23:30:11 -05:00
Gian-Carlo Pascutto
4282195cd5 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
a15e9d5a83 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
47bd0a7bc7 Bug 820550 - Signaling - stop checking API state in CloseInt r=ekr 2012-12-12 09:39:51 -08:00
EKR
1ab56747cb Bug 820671 Make PCImpl a tombstone if mMedia is null r=jesup 2012-12-11 19:45:09 -08:00
Dan Mosedale
1f420ec7bf 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
a73f9fcfe0 bug 820769 - fix mozmake.py to generate correct common.mk include. r=jesup 2012-12-12 11:32:05 -05:00
EKR
fe254a31df Bug 792175 - Move PeerConnection operations onto main thread. r=jesup,derf,ehugg 2012-09-26 10:14:23 -07:00
Gian-Carlo Pascutto
955bc3788f Bug 816575 - Remove call to SetRenderAndroidVM when not using internal renderer. r=tterribe 2012-12-11 20:11:15 +01:00
Gian-Carlo Pascutto
39d4d2f61a 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
ccdfbe4a73 Bug 752657: backout sigslot multithreaded due to deadlocks rs=jesup 2012-12-11 12:54:24 -05:00
Randell Jesup
17c6d540a7 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
8422656a48 Bug 806822: avoid race condition in gathering stats r=ehugg 2012-12-11 07:58:53 -05:00
Steven Lee
d777fcb4f5 Bug 819246 - arm_neon is meaningful only when armv7 is true. r=jesup 2012-12-10 09:05:00 -05:00
Ed Morley
8c15e9f014 Merge mozilla-central to mozilla-inbound 2012-12-10 14:03:11 +00:00
Mike Hommey
e94416d6cb 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
34b6e285e3 Bug 816112: Configure webrtc video/RTCP parameters for better defaults r=jesup 2012-11-28 08:34:10 -08:00
Oleg Romashin
43b01abc72 Bug 819364 - Webrtc signaling unit test does not build on Qt port. r=rjesup 2012-12-09 09:28:37 -08:00
Saurabh Anand
c9e3659e16 Bug 818817 - Fix some compiler warnings, r=Ms2ger 2012-12-09 22:53:19 +05:30
Adam Roach [:abr]
76556b50cb 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
86d868d420 Bug 817488 Audio Send and Recv support for PC tests r=ekr 2012-12-05 13:37:32 -08:00
Mike Hommey
03734a3cb2 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
ecd3a4e29c Bug 814329 Protect fim_process_event from NULL input r=jesup 2012-11-28 08:38:50 -08:00
Gian-Carlo Pascutto
791f727b6d 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
32394d35a3 Bug 817481 - Build failure on WebRTC unit test with --with-system-libvpx. r=jesup 2012-12-03 17:01:16 +09:00
Crypt
ab8161fb27 Bug 810220 - Patch to fix SDP Codec Negotiation Issues (revised) r=ekr,jesup 2012-11-26 02:36:43 -08:00
EKR
f1462ce7dd Bug 817451, Correctly set mRole in PC unit tests. r=ehugg 2012-12-02 18:06:21 -08:00
Brian Smith
d935774988 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
94e8c57fdc Bug 812636: additional checks for failure for a Conduit be created r=ehugg 2012-11-28 03:28:30 -05:00
Daniel Holbert
1f6c42c110 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
08944b1306 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
b6cd1fb640 Bug 814734 - Fixed Log format string mismatches r=ekr 2012-11-26 08:38:14 -08:00
EKR
11e2f21d47 Bug 805662, Crash in mediaconduit_unittest. r=jesup 2012-11-22 10:57:37 -08:00
Adam Roach
bdc8045c11 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
3b241cb7e1 Bug 813746. Link the webrtc unit tests to ZLIB as needed. r=bz 2012-11-21 02:16:03 -05:00
Ethan Hugg
3013b2da47 Bug 813723 Reorder state assignment in unit tests r=ekr 2012-11-20 13:47:07 -08:00
Ethan Hugg
7c0b8a666a Bug 811118 build webrtc unittests by default but run only some r=jesup 2012-11-12 15:34:05 -08:00
Ethan Hugg
e7c5bd04af Bug 813212 Update signaling unittests to match latest IDL r=jesup 2012-11-19 09:59:36 -08:00
EKR
e0a0253926 Bug 799419: Force NSS startup during PeerConnection Initialize r=bsmith,jesup 2012-11-18 07:43:26 -08:00
Randell Jesup
645ded6229 Bug 812886: Watch network (tear)down events and kill PeerConnections r=ekr,smaug,bsmith 2012-11-18 23:53:14 -05:00
Randell Jesup
c4881797fc Bug 812641: Shut down SipCC instance when number of PeerConnections == 0 (reland) r=ehugg 2012-11-17 23:03:58 -05:00
Randell Jesup
7b5d421611 Bug 812641: Shut down SipCC instance when number of PeerConnections==0 r=ehugg 2012-11-18 01:42:40 -05:00
Ryan VanderMeulen
00898e7ef8 Backed out changeset b527670b6728 (bug 811118) for orange. 2012-11-16 21:24:14 -05:00
Nathan Froyd
014d437d3a 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
35febb02ed Bug 811118 build webrtc unittests by default r=jesup 2012-11-12 15:34:05 -08:00
Randell Jesup
02d84ce8c8 Bug 806830: Enforce initializing strlib before using r=ehugg 2012-11-16 14:37:08 -05:00
EKR
5cf9bf0408 Bug 811183: Recursive GC In PeerConnection shutdown; r=jesup 2012-11-16 10:27:30 -08:00
Timothy B. Terriberry
7a1d1f66f1 Bug 810363 - Reject non-stereo, non-48 kHz Opus streams, r=ehugg 2012-11-15 15:09:39 -08:00
Timothy B. Terriberry
d8713fd2b5 Bug 810353 - Offer Opus as stereo instead of mono, r=ehugg 2012-11-15 15:09:39 -08:00
Randell Jesup
4e788c410e Bug 811695: disable internal socket transports for getUserMedia Audio capture r=derf 2012-11-15 17:58:40 -05:00
Adam Roach
a19c8d14bd Bug 803318: Improved handling of constraints and more tests. r=ekr 2012-11-14 11:25:14 -06:00
Ethan Hugg
42e87828fc Bug 811333 Signaling - Fix size of thread ID for Windows build r=jesup 2012-11-13 09:23:04 -08:00
Ethan Hugg
f794a0b56e Bug 806407 Fix type in struct passed into msgsnd f=jesup 2012-11-09 08:47:00 -08:00
Nathan Froyd
ce7eadda0a Bug 809950 - fix webrtc signaling Wrapper code to not introduce static initializers; r=jesup 2012-11-08 13:45:39 -05:00
Randell Jesup
5f9b147f0d Bug 807929: Make DataChannel refcounted r=mcmanus 2012-11-02 15:28:13 -04:00
EKR
f00d9d3eab Bug 807259: Fix shutdown for WebRTC standalone unit tests. r=anant 2012-10-31 16:31:47 +01:00
Randell Jesup
aeec63cf4b Bug 805279: make sure we pass values for formatted log messages r=derf 2012-10-30 15:31:09 -04:00
EKR
7aef558c74 Bug 806335: MediaPipeline destroys TransportFlow on wrong thread; r=anant 2012-10-29 15:35:12 +01:00
EKR
3c40009bf3 Bug 806306: Fix compile bustage for MediaConduit; r=anant 2012-10-29 15:33:19 +01:00
Robert O'Callahan
8fe699d05c Bug 805703. Part 1: MediaStreamGraph::CreateInputStream -> CreateSourceStream. r=jesup 
--HG--
extra : rebase_source : 3c327e9ead92508f12df4b95f2fd24fa2ba97ab5
 2012-10-29 17:36:31 +13:00
Randell Jesup
20c57e322f Bug 805701: protect against NULL fcb pointer r=ehugg 2012-10-27 21:47:39 -04:00
EKR
74b119ca7d Bug 801221: Move mtransport operations onto STS thread; r=jesup 2012-10-18 13:01:52 -07:00
Ethan Hugg
fdfb2b183d Bug 805533 Send SDP parse errors through single function for reporting r=jesup 2012-10-26 13:05:50 -07:00
Robert O'Callahan
4dcad1fa98 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
b1f3765e26 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
9542499e15 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
40f09ef25e Merge the last PGO-green inbound chnageset to m-c. 2012-10-25 21:14:50 -04:00
Randell Jesup
e289cb5b07 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
6597ec75b3 Bug 803744 - Allow signaling startup when already started r=jesup 2012-10-24 15:46:40 -07:00
Anant Narayanan
70fbb3f665 Bug 802694: Pass along constraints from PC JS module to PCImpl; r=ekr,jesup 2012-10-25 12:24:30 -07:00
Ehsan Akhgari
f117c7a7b0 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
48b5c1a608 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
ea0b9a0c76 Bug 803881: Fix wrong number of samples for webrtc media unit tests r=ekr 2012-10-24 16:54:33 -04:00
Ethan Hugg
25ee34c4f3 Bug 729541 fix for syntax error in signaling_unittests r=jesup 2012-10-23 21:26:27 -07:00
Ethan Hugg
8c014a3816 Bug 797512 Signaling: simplify set_dtls_fingerprint r=jesup 2012-10-22 16:38:52 -07:00
Ethan Hugg
1aa952bdb9 Bug 803875 MediaConstraints - fix length handling r=jesup 2012-10-22 13:25:44 -07:00
Enda Mannion
0ba819af9f Bug 729541: Remove 'offer' parameter from RTCPeerConnection.createAnswer r=jesup 2012-10-15 16:50:10 +01:00
Randell Jesup
7b88f6bbd8 Bug 803086: Process NULL image chunks in NotifyQueuedTrackChanges() r=ekr 2012-10-19 10:58:06 -04:00
Enda Mannion
19a1757572 Bug 800688: remove local_dynamic_payload_type_value from negotiation r=ehugg,jesup 2012-10-17 15:57:52 +01:00
Ethan Hugg
bd07838607 Bug 798873 Patch 3 - flex_string fix for Windows vsnprintf r=jesup 2012-10-17 18:57:57 -07:00
Mike Hommey
d964501004 Bug 799975 - Fix webrtc when building with system nspr/nss. r=ted 2012-10-17 16:33:23 +02:00
Mike Hommey
99b86d0934 Bug 798926 - Define INTEGER_TYPES_H and others for srtp.h. r=jesup 2012-10-17 16:28:33 +02:00
Enda Mannion
5a8e92c267 Bug 786152: Increment DataChannel SCTP port in a WebRTC JSEP answer r=jesup 2012-09-28 11:36:02 +01:00
Enda Mannion
33296357c8 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
1735393652 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
181fa73bef 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
3d71f52bfc Bug 800611 - Trailing whitepace removed from signaling code r=jesup 2012-10-12 08:15:24 -07:00
Gervase Markham
1da1c4ba1e Bug 796457 - upgrade license to MPL 2. 2012-10-12 15:58:11 +01:00
Ed Morley
8fc88262ba 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
67399d2474 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
df9438058b Bug 800343 - Cast nsresult to uint32_t to output to streams; r=ehsan 2012-10-11 14:48:31 +02:00
EKR
e23c5f65b3 Bug 792811 - Check creation of WebRTC transportflows r=jesup 2012-10-09 12:29:01 -07:00
Ethan Hugg
76eb10e75c Bug 800502 - Signaling - Removed unused code that required XML handling r=jesup 2012-10-11 14:07:50 -07:00
Ethan Hugg
cf930efa7b Bug 800558 - Signaling - return proper error code in fsmdef.c r=jesup 2012-10-11 14:32:45 -07:00
Ethan Hugg
94d4df0c19 Bug 798873 - Signaling SDP construction uses flex_string r=jesup 2012-10-08 16:43:33 -07:00
Gregory Szorc
5b895d9902 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
7b4b67c803 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
b3ea7773b7 Backout d6e84a64d3d4 (bug 798969) for burning 2012-10-11 14:37:44 +01:00
Robert O'Callahan
062d113c60 Bug 798969. Handle symlinked srcdirs by avoiding use of Python's __file__. r=jesup 2012-10-08 17:45:10 +13:00
Ethan Hugg
7127744548 Bug 791278: Protect PeerConnection setLocal/RemoteDescription from NULL input r=jesup 2012-10-09 15:14:51 -07:00
Ethan Hugg
eb386b8884 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
bc7bf01608 Bug 799477: missing stdarg.h in CSFLog.h (typically gcc 4.4.x) r=ehugg 2012-10-09 14:46:34 -04:00
EKR
9bcb94d7e2 Bug 799246: Conditionally enable webrtc unit tests r=jesup,ted,cjones 2012-10-08 18:56:00 -07:00
Randell Jesup
23a78a8528 Bug 799071: clean up clang warnings in media/webrtc/signaling (-Werror) r=ehugg 2012-10-08 17:11:10 -04:00
Ethan Hugg
25aeed0845 Bug 798873 - Signaling - increase max SDP size (temporary until better fix). r=jesup,ekr 2012-10-07 22:28:52 -07:00
Randell Jesup
7569f2e9db 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
de7cb6110e 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
c284b98070 Bug 579517 follow-up: Remove NSPR types that crept in 2012-10-07 18:26:08 -04:00
Anant Narayanan
0e8bef661b Bug 798825: Add DataChannel DOM interfaces to RTCPeerConnection; r=smaug 2012-10-07 01:34:30 -04:00
Anant Narayanan
785ec1a331 Bug 694807: Implement PeerConnection C++ module; r=jst,jesup,ekr 2012-10-07 01:34:30 -04:00
EKR
2ae69bfc6d 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
9d2502d0a4 Bug 792188: Fixes to signaling MediaPipeline and Srtpflow r=jesup,derf 2012-10-07 01:34:29 -04:00
Randell Jesup
82c91f1b08 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
09f2e4a618 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
a16c4614b0 Bug 792188: Import SIPCC into media/webrtc/signaling rs=jesup r=gerv 2012-03-13 11:22:14 -07:00
Randell Jesup
123a8599dc 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
85a14018f1 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
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rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/standard/video_sync_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/standard/video_sync_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/standard/voe_base_misc_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/standard/volume_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_cpu_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_cpu_test.h
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_extended_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_extended_test.h
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_standard_test.h
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_stress_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_stress_test.h
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_test_defines.h
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_unit_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h => media/webrtc/trunk/src/voice_engine/test/auto_test/voe_unit_test.h
rename : media/webrtc/trunk/src/voice_engine/main/test/cmd_test/Android.mk => media/webrtc/trunk/src/voice_engine/test/cmd_test/Android.mk
rename : media/webrtc/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc => media/webrtc/trunk/src/voice_engine/test/cmd_test/voe_cmd_test.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/voice_engine_tests.gypi => media/webrtc/trunk/src/voice_engine/test/voice_engine_tests.gypi
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/Resource.h => media/webrtc/trunk/src/voice_engine/test/win_test/Resource.h
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTest.aps => media/webrtc/trunk/src/voice_engine/test/win_test/WinTest.aps
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTest.cc => media/webrtc/trunk/src/voice_engine/test/win_test/WinTest.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTest.h => media/webrtc/trunk/src/voice_engine/test/win_test/WinTest.h
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTest.rc => media/webrtc/trunk/src/voice_engine/test/win_test/WinTest.rc
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc => media/webrtc/trunk/src/voice_engine/test/win_test/WinTestDlg.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h => media/webrtc/trunk/src/voice_engine/test/win_test/WinTestDlg.h
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico => media/webrtc/trunk/src/voice_engine/test/win_test/res/WinTest.ico
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2 => media/webrtc/trunk/src/voice_engine/test/win_test/res/WinTest.rc2
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/stdafx.cc => media/webrtc/trunk/src/voice_engine/test/win_test/stdafx.cc
rename : media/webrtc/trunk/src/voice_engine/main/test/win_test/stdafx.h => media/webrtc/trunk/src/voice_engine/test/win_test/stdafx.h
rename : media/webrtc/trunk/src/voice_engine/main/source/transmit_mixer.cc => media/webrtc/trunk/src/voice_engine/transmit_mixer.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/transmit_mixer.h => media/webrtc/trunk/src/voice_engine/transmit_mixer.h
rename : media/webrtc/trunk/src/voice_engine/main/source/utility.cc => media/webrtc/trunk/src/voice_engine/utility.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/utility.h => media/webrtc/trunk/src/voice_engine/utility.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc => media/webrtc/trunk/src/voice_engine/voe_audio_processing_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h => media/webrtc/trunk/src/voice_engine/voe_audio_processing_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_base_impl.cc => media/webrtc/trunk/src/voice_engine/voe_base_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_base_impl.h => media/webrtc/trunk/src/voice_engine/voe_base_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_call_report_impl.cc => media/webrtc/trunk/src/voice_engine/voe_call_report_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_call_report_impl.h => media/webrtc/trunk/src/voice_engine/voe_call_report_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_codec_impl.cc => media/webrtc/trunk/src/voice_engine/voe_codec_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_codec_impl.h => media/webrtc/trunk/src/voice_engine/voe_codec_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc => media/webrtc/trunk/src/voice_engine/voe_dtmf_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_dtmf_impl.h => media/webrtc/trunk/src/voice_engine/voe_dtmf_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_encryption_impl.cc => media/webrtc/trunk/src/voice_engine/voe_encryption_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_encryption_impl.h => media/webrtc/trunk/src/voice_engine/voe_encryption_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_external_media_impl.cc => media/webrtc/trunk/src/voice_engine/voe_external_media_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_external_media_impl.h => media/webrtc/trunk/src/voice_engine/voe_external_media_impl.h
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_file_impl.cc => media/webrtc/trunk/src/voice_engine/voe_file_impl.cc
rename : media/webrtc/trunk/src/voice_engine/main/source/voe_file_impl.h => media/webrtc/trunk/src/voice_engine/voe_file_impl.h
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
foudfou
d72827c0aa 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
2535e26cb5 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
d1bb662c3c 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
6173fa297f Bug 774032 bonus - Use @DEPTH@ and @relativesrcdir@ in Makefile.in. r=ted 2012-08-04 20:26:44 +02:00
Andrew Benton
dec4441699 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
7981252808 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
d9801ef36c 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
3bcf9c0162 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
06770ac1e7 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
ce54a40fdb 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
bdd3a9d31d 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
ca925c1bd9 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
04cb98a9a8 Bug 757637: Rollup makesystem changes for webrtc r=khuey r=ted f=glandium 2012-06-20 07:27:43 -04:00
Randell Jesup
9f2035f2a4 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
8eeefb2528 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
cd18b40464 bug 731407: need to include <assert.h> in non-debug builds r=cpearce 2012-04-04 14:49:12 -04:00
Bas Schouten
e24d1ed6c4 bug 731407: Remove DShow BaseClass usage from webrtc drop r=cpearce 2012-03-15 23:06:35 +00:00
Randell Jesup
2117ab705d 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