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Author SHA1 Message Date
leozwang@webrtc.org
425e680808 Enable PLI as the default. 
Description:
Enable PLI as the default option.

BUG=webrtc issue 744
TEST=local
Review URL: https://webrtc-codereview.appspot.com/735008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-14 17:03:33 +00:00
mflodman@webrtc.org
90071dd647 Added API to set RTP timestamp offset extension. 
BUG=745

Review URL: https://webrtc-codereview.appspot.com/710011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:13:27 +00:00
mflodman@webrtc.org
1fb39ba422 REMB changes, cloned from issue 722011. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/708012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:05:14 +00:00
leozwang@webrtc.org
a11299648c Retrieve data from input 
Espeically on tablet, we have to read data dirtectly from input text edit rather than
track key input to let text edit get updated automatically

BUT=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/705010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-11 04:33:02 +00:00
andrew@webrtc.org
cdfa63f94f Fix mismatched signature (due to const) error. 
TBR=mikhal@webrtc.org
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/717013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 22:56:17 +00:00
henrike@webrtc.org
7742479428 Fixes build bot breakage. Resizing was enabled which some tests assumed wouldn't be the case. Changed the default so that it is now disabled. 
Review URL: https://webrtc-codereview.appspot.com/731006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 19:31:24 +00:00
henrike@webrtc.org
268a24fa56 Reverts changes to auto test. 
Review URL: https://webrtc-codereview.appspot.com/724006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 15:15:51 +00:00
astor@webrtc.org
c0496e66f6 Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP. 
Review URL: https://webrtc-codereview.appspot.com/678007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 10:14:43 +00:00
henrike@webrtc.org
3c286747ce Makes it possible to disable automatic resizing. 
Review URL: https://webrtc-codereview.appspot.com/710010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 02:44:30 +00:00
leozwang@webrtc.org
b20916e336 Change libaries path because of recent file structure changes 
Description:
1. Changed file path.
2. Because of optimization code changes, a new neon library is created, add it to finial build.

BUG=None
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/731005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-09 22:17:26 +00:00
phoglund@webrtc.org
54e22eb977 Made it possible to run video_capture tests on mac. 
Abstracted out a suitable main from vie_auto_test and put it into testsupport.
Cleaned up unused vie_auto_test mac code.

BUG=

Review URL: https://webrtc-codereview.appspot.com/723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-08 08:27:46 +00:00
mflodman@webrtc.org
1e1a250413 Wrong RTP module used when calling RegisterReceiveRtpHeaderExtension in ViE channel. 
Review URL: https://webrtc-codereview.appspot.com/717010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-06 08:49:41 +00:00
elham@webrtc.org
c0348fb349 bump version to 3.9.0 
Review URL: https://webrtc-codereview.appspot.com/708007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-03 17:47:52 +00:00
andrew@webrtc.org
d7a71d0719 Prepare to roll Chromium to 149181. 
- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-01 01:40:02 +00:00
mikhal@webrtc.org
a2031d58f6 Replacing RawImage with VideoFrame in video_coding and related engine code. 
This is the first step of replacing RawImage with VideoFrame in all WebRtc modules.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/672010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-31 15:53:44 +00:00
andrew@webrtc.org
6f8db36e04 Reorganize voice_engine/. 
The usual changes:
voice_engine/main/source -> voice_engine/
voice_engine/main/interface -> voice_engine/include
voice_engine/main/test -> voice_engine/test
Include path changes.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/705004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 21:49:28 +00:00
andrew@webrtc.org
c1354bd768 Make handling of libyuv more flexible. 
- Use gyp variable for libyuv path.
- Rename internal libyuv.h to webrtc_libyuv.h to avoid conflicts.
- Update affected includes.

BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/711004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 18:21:16 +00:00
mikhal@webrtc.org
7cbb5a05c4 JPEG: Replacing RawImage with VideoFrame. 
Replacing RawImage with VideoFrame in JPEG related code

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/703004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-25 20:38:14 +00:00
leozwang@webrtc.org
8d95a700e9 Change libvp8 library patch in makefile 
It's caused by recent file structure changes in vp8

TBR=ronghua,kma
BUG=
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/707004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2528 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-25 18:37:00 +00:00
andrew@webrtc.org
f5a91fdfab Make some build settings more flexible. 
BUG=issue676
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/700006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-23 16:28:02 +00:00
leozwang@webrtc.org
8495915442 Make loopback mode work properly 
Some minor changes and improvements are added into this cl

BUG=
TEST=vie_test
Review URL: https://webrtc-codereview.appspot.com/667005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-16 20:03:18 +00:00
mikhal@webrtc.org
73db8dbfc2 video conversion functions: switching from designated functions to a general one. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/686004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-14 00:03:55 +00:00
leozwang@webrtc.org
7760963d04 Make webrtc compile on android in chromium 
Message:
There probably is a better way, this cl is trying to seperate android
specific calls into android files, particular SetAndroidObject, by doing
this, webrtc can be built inside Chromium on android. Currently, Chromium
manages its own jvm, capturer and renderer, all webrtc code that manages
jvm, captuer and renderer should not be compiled. 

Description:
By re-organize android specific code, this cl will make webrtc build
in Chromium on android.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/668007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 22:00:43 +00:00
leozwang@webrtc.org
6c08f26c4e Terminate version string 
This cl doesn't directly solve b/6750185, but it's a potential bug
if string is not terminated correctly

BUG=
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/674009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 22:00:16 +00:00
marpan@webrtc.org
71707aaae8 Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 
Review URL: https://webrtc-codereview.appspot.com/682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 16:27:51 +00:00
leozwang@webrtc.org
cf9855d9eb Update build.xml and api level 
Description:
This cl updates build.xml following the sdk_r20 release. Also upgrade api
level to 10. API level 9 is obsolete and we don't reply on level 9 particular
features, upgrade to 10 to make development more easier.

BUG=
TEST=local build
Review URL: https://webrtc-codereview.appspot.com/678005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-10 17:38:48 +00:00
stefan@webrtc.org
ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset 
- Change how the transmission offset is calculated, to
  incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
  We must use the same clock as in the RTP module to be able to measure
  the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/666006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 13:21:22 +00:00
vikasmarwaha@webrtc.org
e85c77bd7c Bump WebRTC version to 3.8.1 
Review URL: https://webrtc-codereview.appspot.com/665007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-02 18:11:06 +00:00
leozwang@webrtc.org
ea5b8b5903 Trival changes in gui layout based on feedback 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/674006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 18:31:45 +00:00
mflodman@webrtc.org
e06ca3cef6 Removed nolint for include guards. 
BUG=
TEST=cpplint.py --filter=-build/header_guard src/video_engine

Review URL: https://webrtc-codereview.appspot.com/676008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 13:20:14 +00:00
mflodman@webrtc.org
ab2610ffd9 Removed the last lint warnings in video_engine. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/670006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-29 10:05:28 +00:00
mflodman@webrtc.org
c802e0ed0c Changed max codec resolution. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/674008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:57:39 +00:00
stefan@webrtc.org
5f28498149 First step in refactoring audio/video synchronization. Adds unittests. 
BUG=
TEST=stream_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:51:16 +00:00
mflodman@webrtc.org
cee447a5bb cpplint passes for vie_performance_monitor, vie_manager_base, vie_impl, vie_renderer, vie_defines and vie_render_manager. 
NOLINT is used where API changes would be needed, for include guards and include files in WebRTC root.

Lots of changes, but no real logical changes.

BUG=627
TEST=vie_auto_test + compiles on all platforms.

Review URL: https://webrtc-codereview.appspot.com/679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:29:46 +00:00
asapersson@webrtc.org
100463e828 Added initial nack configuration for rtp module. 
Review URL: https://webrtc-codereview.appspot.com/677007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:21:51 +00:00
mflodman@webrtc.org
1b1cd78dd2 Made cpplint pass for vie_remb, vie_ref_count, vie_sender and vie_receiver. 
NOLINT is used for include guards. I took a shortcut for vie_ref_count, the class will be deleted very soon anyway.

BUG=627
TEST=cpplint and compiles

Review URL: https://webrtc-codereview.appspot.com/677008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 06:34:08 +00:00
mflodman@webrtc.org
f5e99db10b Made cpplint pass for vie_channel.* and vie_encoder.*. NOLINT is used for API changes, include guards and include files in WebRTC root. 
WebRTC types and webrtc:: will be removed in a follow up.

BUG=627
TEST=vie_auto_test + compiles

Review URL: https://webrtc-codereview.appspot.com/677005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 09:49:37 +00:00
andrew@webrtc.org
81cf5e4752 Move test to src/test. 
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing.
- Remove now unneeded third_party_root.

TBR=henrike@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 01:41:54 +00:00
astor@webrtc.org
bd7aeba8fb Expose a set of options to the OveruseDetector supporting experiments 
Updated overuse_detector.* to use google style naming convention
Removed OveruseDetector::Reset
Review URL: https://webrtc-codereview.appspot.com/666005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-26 10:47:04 +00:00
henrike@webrtc.org
643be71700 Adds variable for third party directory. 
BUG=348
TEST=Manual testing in Chrome and WebRTC workspace.

Review URL: https://webrtc-codereview.appspot.com/674005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 10:48:58 +00:00
mflodman@webrtc.org
64f86fba19 Fix test app render bug. 
Review URL: https://webrtc-codereview.appspot.com/669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 12:32:39 +00:00
mflodman@webrtc.org
8baed51f6e This CL is part of enabling cpplint check for video_engine uploads. 
BUG=627
TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/653006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 12:11:50 +00:00
mflodman@webrtc.org
9ba151bdf9 Removed cpplint warnings from all impl-files to be able to add this check as presubmit step. I don't want to change the API right now, will come later, so there are several NOLINT comments added to get around this for now. 
BUG=627
TESTS=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/661005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 10:02:13 +00:00
hta@webrtc.org
2bd8d62d3b Sleep using no compile flags 
BUG=603
TEST=

Review URL: https://webrtc-codereview.appspot.com/665004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 09:57:24 +00:00
mflodman@webrtc.org
67f98ec63a Removed flaky REMB test. This test is now covered by: 
- RemoteBitrateEstimatorTest
- BitrateControllerTest
- RtcpFormatRembTest
- ViERembTest

BUG=477
TEST=See above.

Review URL: https://webrtc-codereview.appspot.com/667004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 09:29:53 +00:00
wu@webrtc.org
2259f855ea Remove unused member variables found by clang's -Wunused-private-field. 
No intended behavior change.

On behavior of thakis@chromium.org.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/641011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 14:56:50 +00:00
vspasova@webrtc.org
f477aac844 Removed gflags header from vie_auto_test. 
Removed gflags include file from src/video_engine/test/automated/
vie_video_verification.cc as it is no longer needed.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/645005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 09:20:33 +00:00
mflodman@webrtc.org
e3a0712f04 Deregister RTP module before deleting it. 
BUG=617
TEST=

Review URL: https://webrtc-codereview.appspot.com/661004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 12:43:41 +00:00
niklas.enbom@webrtc.org
d63d06a289 bump version to 3.8 
Review URL: https://webrtc-codereview.appspot.com/657004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 08:36:36 +00:00
mflodman@webrtc.org
139c4678c1 Fixed a/v sync issue. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 11:08:51 +00:00