Commit Graph

3839 Commits

Author SHA1 Message Date
henrike@webrtc.org
9de257d00f Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1824004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-17 14:42:53 +00:00
pbos@webrtc.org
a3f30143b7 Default constructor for RtcpAppHandler.
Whenever this test (RtcpApplicationDefinedPacketsCanBeSentAndReceived) fails
because it's being run on a slower system (such as one running under valgrind),
valgrind reports a lot of undefined-value errors. Initializing the data
makes sure that, while the EXPECT_EQs trigger, they don't cause any errors in
valgrind.

BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4363 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-17 14:25:45 +00:00
tnakamura@webrtc.org
64e2cbf184 clean up incomplete revert in r4357
Also revert r4319, will follow up with pbos

Reason for recent series of reverts: video freezes when testing with packet loss

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 21:52:59 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
henrike@webrtc.org
7b2f955e56 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.
BUG=1935
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1805005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
henrike@webrtc.org
42581545eb Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
Libjingle sets the variables here: https://code.google.com/p/libjingle/source/browse/trunk/talk/supplement.gypi

BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4354 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 16:37:22 +00:00
pbos@webrtc.org
3d8647f17d Include files from webrtc/.. paths in signal_processing/.
BUG=1662
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4352 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:32:03 +00:00
pbos@webrtc.org
0c4e05afbb Include files from webrtc/.. paths in media_file/.
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4351 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:05:40 +00:00
pbos@webrtc.org
9b82dced8d Make sure first RTP packet counts as in-order.
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1811004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:03:35 +00:00
pbos@webrtc.org
2e10b8e4a0 Include files from webrtc/.. paths in bitrate_controller/.
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:54:53 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/.
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
elham@webrtc.org
4a44ea21d7 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc"
TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1803004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:46:06 +00:00
elham@webrtc.org
4888fd4827 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered"
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:21:48 +00:00
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:08:27 +00:00
elham@webrtc.org
6f5707e184 Revert r4328
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 20:59:52 +00:00
elham@webrtc.org
8543c1c77c Updated WebRTC version to 3.36
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1780005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4341 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 17:19:45 +00:00
marpan@webrtc.org
ca35c19e5a Roll libvpx to 208227.
-pick up libvpx roll to 93f88ab.

TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/1798004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4340 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 21:08:26 +00:00
pbos@webrtc.org
df119c9a45 Remove dead video_capture for QuickTime.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4339 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 18:08:13 +00:00
henrike@webrtc.org
723d683ecb Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1797004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 16:04:50 +00:00
pbos@webrtc.org
a9b74ad716 Include files from webrtc/.. paths in video_capture/.
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4337 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 10:03:52 +00:00
pbos@webrtc.org
8b06200802 Include files from webrtc/.. paths in utility/.
BUG=1662
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:28:10 +00:00
pbos@webrtc.org
0ed57c51a3 Remove dead code testAPI.cc.
BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4335 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:23:05 +00:00
pbos@webrtc.org
5aa3f1b4c0 Include files from webrtc/.. paths in video_render/.
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4334 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:12:08 +00:00
pbos@webrtc.org
5b10d8fb18 Fix some voe_auto_test uninitialised-value errors.
BUG=
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4332 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 15:50:07 +00:00
henrike@webrtc.org
ffe16bdae9 trunk/talk: removes empty folders.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4331 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 15:42:10 +00:00
pbos@webrtc.org
811269df40 Include files from webrtc/.. paths in audio_device/.
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 13:24:38 +00:00
pbos@webrtc.org
db6e3f8bc5 Fix root-relative includes for pacing/.
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4329 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 09:50:05 +00:00
stefan@webrtc.org
e4736eee20 Fixes a crash when sending SR reports from a sender only module.
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 08:28:35 +00:00
braveyao@webrtc.org
aeba6e8740 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API.
BUG=2051
TEST=autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 08:06:37 +00:00
pbos@webrtc.org
96edd56170 Sorted headers under rtp_rtcp/.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 15:40:42 +00:00
pbos@webrtc.org
69215d8432 Include files from webrtc/.. paths in video_engine/.
BUG=1662
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1759005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 15:02:02 +00:00
pbos@webrtc.org
adf23a55f8 Direct3D renderer for new VideoEngine API tests.
TEST=Rendered video in video_loopback test.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1573004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 14:07:56 +00:00
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs.
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 13:39:27 +00:00
stefan@webrtc.org
9de89a6f6b Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered.
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 12:42:15 +00:00
stefan@webrtc.org
452d853c43 Fix three uninitialized members in rtp_receiver_impl.cc.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 10:54:56 +00:00
pbos@webrtc.org
08933a5dfb Initialize payload-type frequency in channel.cc.
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.

BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1780004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 10:06:29 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
tnakamura@webrtc.org
6aa6229953 Update version number to 3.35
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1778004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4316 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 18:43:02 +00:00
tnakamura@webrtc.org
c79b9295cd Update version number to 3.34
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1770006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4315 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 18:40:52 +00:00
pbos@webrtc.org
fc496d95df Add root_path_android.cc to webrtc/test/Android.mk.
Fixes the broken android-platform build (build that uses .mk files).

TBR=andrew@webrtc.org,henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1777004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4314 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 15:24:16 +00:00
pbos@webrtc.org
f3f1358360 Fixed implicit-int-conversion bugs.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 14:04:46 +00:00
stefan@webrtc.org
cab716cc7d Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
TBR=henrikg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 13:43:24 +00:00
stefan@webrtc.org
f56d612c70 Create gyp target for bwe components.
R=henrikg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1775004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 12:32:35 +00:00
pbos@webrtc.org
af8d5afec9 Initial port of FullStackTest to new VideoEngine API.
Deferring network loss, delay and such to a later CL.

BUG=1872
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1756004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4310 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 08:02:33 +00:00
henrike@webrtc.org
5fc4d34f54 Arguments need to be separated when implementing gyp-actions.
TBR=andrew@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1774004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 02:08:25 +00:00
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:31:18 +00:00
henrike@webrtc.org
e80a934b36 Added modules_unittests.isolate for ndk-apk builds.
TBR=csharp@chromium.org, frankf@chromium.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1750004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4307 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:19:57 +00:00
henrike@webrtc.org
a950300b0e Disables unit tests that don't work on Android for Android.
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 18:53:54 +00:00
henrike@webrtc.org
a2073af728 Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1770004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4305 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 18:14:58 +00:00
henrike@webrtc.org
bd3eee3e24 Fixes broken gyp-condition.
TBR=andrew@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1771004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4304 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 17:34:20 +00:00