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 |
|
henrike@webrtc.org
|
34773d9b6b
|
Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests.
TBR=andrew@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1754005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4303 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-08 14:55:23 +00:00 |
|
pbos@webrtc.org
|
1932fe1865
|
Use scoped_ptr<> for loopback.cc
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1764004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4302 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-05 17:02:37 +00:00 |
|
stefan@webrtc.org
|
66b2e5c05a
|
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-05 14:30:48 +00:00 |
|
mcasas@webrtc.org
|
d4d9480c05
|
Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4300 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-05 09:12:04 +00:00 |
|
pbos@webrtc.org
|
db7d82f26f
|
Revert 4298 "Makes it possible to find files used by some unit t..."
> Makes it possible to find files used by some unit tests when running them as Chrome native tests.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1749004
Broke Android NDK/Android.mk builds.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1752006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4299 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-05 08:49:09 +00:00 |
|
henrike@webrtc.org
|
caf2fcca6a
|
Makes it possible to find files used by some unit tests when running them as Chrome native tests.
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4298 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-05 04:15:38 +00:00 |
|
mflodman@webrtc.org
|
21beaf97e7
|
Adding Stefan as VideoEngine owner, removing Per.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1762004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4296 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-04 12:29:08 +00:00 |
|
braveyao@webrtc.org
|
0b8636a783
|
In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed.
BUG=
TEST=manual Test
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1753005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4295 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-04 07:24:12 +00:00 |
|