367 Commits

Author SHA1 Message Date
sergeyu@chromium.org
9e182795a9 Enable ScreenCapturer unittests
previously ScreenCapturer unittests were disabled by mistake

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4238 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 21:14:36 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests.
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 20:09:44 +00:00
kjellander@webrtc.org
6c35e0b0f7 Reorganize test targets in WebRTC
This CL will lower the number of test targets in WebRTC by:

Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests

Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests

Merge into test_support_unittests:
* channel_transport_unittests

channel_transport.gyp was also removed in favor for test.gyp.

I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.

Buildbot configuration update will be synced with the commit of this CL.

TEST=trybots
BUG=1843
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11 08:29:17 +00:00
kjellander@webrtc.org
fec34d7afa Merge webrtc_utility_unittests into modules_unittests.
This CL eliminates the webrtc_utility_unittests test target.

NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.

BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1584004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 08:58:46 +00:00
sergeyu@chromium.org
15e32ccd30 Add DesktopCapturer interface for desktop capturers.
The new DesktopCapturer interface will be used for screen and window
captures. Beside DesktopCapturer itself also added classes/interfaces
that it depends on.

R=alexeypa@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1322007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3917 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-29 20:10:57 +00:00
kjellander@webrtc.org
1b427719dc emove desktop_capture.gypi from modules.gyp
When adding this in
we started getting linking problems on the mac_asan bot due to
the empty list of source files for the library target.

Please re-add it into modules.gyp when the library has source files
to compile.

BUG=none
TEST=Passing mac_asan trybot.

Review URL: https://webrtc-codereview.appspot.com/1313009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3882 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-22 13:13:49 +00:00
sergeyu@chromium.org
a0cd9182aa Add desktop_capture directory for screen and window capturers.
The screen captures will be moved from chromium to WebRTC to make it easy
to share this code with other projects. This CL adds a new directory where
the current screen capturer code will be moved to.

Review URL: https://webrtc-codereview.appspot.com/1297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3866 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-17 18:21:42 +00:00
stefan@webrtc.org
7bc465bd21 Fix issues with incorrect wrap checks when having big buffers and high bitrate.
Introduces shared functions for timestamp and sequence number wrap checks.

BUG=1607
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1291005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 17:48:02 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 23:25:25 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 17:52:42 +00:00
henrik.lundin@webrtc.org
d94659dc27 Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.

It has been through extensive internal review during the course of
the project.

TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1073005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 12:09:21 +00:00
stefan@webrtc.org
8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module.
Review URL: https://webrtc-codereview.appspot.com/934023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 14:01:46 +00:00
pwestin@webrtc.org
b518017e71 Adding pacing module, will replace the transmission_bucket in the RTP module.
TESTED=unittest
Review URL: https://webrtc-codereview.appspot.com/930015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3073 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 20:56:23 +00:00
andrew@webrtc.org
9841d92b8d Reorganize modules/video_render.
The usual elimination of main/source etc.

Review URL: https://webrtc-codereview.appspot.com/929011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3027 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-31 05:22:11 +00:00
andrew@webrtc.org
94caca7105 Reorganize modules/video_capture.
- The usual elimination of main/source etc.
- Fully qualify "public" includes.
- Use lowercase for all directory names.

Review URL: https://webrtc-codereview.appspot.com/933010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3025 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 21:58:00 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00