generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
3,652 Commits 1 Branch 0 Tags 108 MiB
4213633a4d
Commit Graph

38 Commits

Author SHA1 Message Date
phoglund@webrtc.org
14d7700d00 Moved command line parsing to internal tools and moved back the mic volume thingie. 
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1491004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 11:52:08 +00:00
phoglund@webrtc.org
5c1948dfaf Moved force_volume_max to its own gyp file to avoid a circular dependency. 
BUG=
TBR=tlegrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 13:59:19 +00:00
andresp@webrtc.org
6b68c28cb1 Added a config class to ease passing a set of options across webrtc. 
Its main design reason is to expose control of experimental webrtc features.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 08:06:36 +00:00
andrew@webrtc.org
c1ffd337f1 Add trace printouts to all unit tests. 
Unfortunately, this requires splitting system_wrappers_unittests out of system_wrappers.gyp to avoid a cyclic dependency.

TESTED=ran a few unit tests and observed printouts

Review URL: https://webrtc-codereview.appspot.com/1221006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:23 +00:00
pwestin@webrtc.org
999e900fb6 Creating a copy of Udp transport under webrtc/test 
Adding a test namespace, updating the include paths and renamed folder name.
Review URL: https://webrtc-codereview.appspot.com/1203004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 16:38:05 +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
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
vspasova@webrtc.org
f61dc9be41 Moving frame_analyzer and rgba_to_i420_converter to src/tools. 
It might be useful to have these under src/tools as this way they will automatically sync in Chrome.

BUG=

Review URL: https://webrtc-codereview.appspot.com/740004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 08:12:00 +00:00
andrew@webrtc.org
54da26d2d2 Move include_tests to a higher variable nesting. 
TBR=wu@webrtc.org
BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/734004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 00:41:18 +00:00
andrew@webrtc.org
cb53410877 Make some dependencies more flexible. 
BUG=none
TEST=trybot

Review URL: https://webrtc-codereview.appspot.com/728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-09 18:28:40 +00:00
andrew@webrtc.org
ea5d0e5162 Don't target top-level test gyps if include_tests==0 
Also add the possibility to pass a flag to the gyp run during gclient hooks.

Review URL: https://webrtc-codereview.appspot.com/729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-09 17:37:03 +00:00
andrew@webrtc.org
d1f3b1a113 Reorganize the vp8 directory. 
The usual changes:
vp8/main/source -> vp8/
vp8/main/test -> vp8/test
vp8/main/interface -> vp8/include

All include paths etc. updated as needed.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/704004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 22:19:16 +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
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
andrew@webrtc.org
b06db96840 Add a framework for audio end-to-end quality testing. 
The quality comparison step is still to be done.

BUG=issue502
TEST=manual

Review URL: https://webrtc-codereview.appspot.com/577005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 18:45:11 +00:00
wu@webrtc.org
caef50310a Removing PeerConnection sample client and libjingle from webrtc. 
The new PeerConnection and sample client can be found in libjingle.
http://code.google.com/p/libjingle/

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 19:23:17 +00:00
kjellander@webrtc.org
cc33737a80 Changing all PSNR/SSIM calculations to use libyuv. 
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.

BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/333025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 08:09:32 +00:00
andrew@webrtc.org
bbea716117 Workaround for libyuv libjingle breakage. 
libjingle depends on ConvertFromI420. This was previously available
through vplib. libjingle still has access to the vplib header, but the
implementation is no longer built.

Fortunately, the libyuv wrapper can supply the implementation, if we
hack the signature to return to the unsigned int types. We'll remove
this once libjingle has been updated to use libyuv directly.

Also, roll libyuv to r100 which fixes a gyp warning on Windows.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 19:43:12 +00:00
henrikg@webrtc.org
af225d6bf6 The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi. 
Review URL: http://webrtc-codereview.appspot.com/315002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 09:58:39 +00:00
andrew@webrtc.org
7fb5d46d3a Give peerconnection its own gyp and disable. 
r1140 broke the libjingle revision we're pulling. The fix in libjingle
is pending; rather than reverting r1140, this temporarily disables
peerconnection in the default build.

TBR=tommi@webrtc.org
TEST=build

Review URL: http://webrtc-codereview.appspot.com/323002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 03:31:41 +00:00
kjellander@webrtc.org
82d91ae6cf Fixing crash when calculating SSIM and PSNR with empty video files in video_metrics.cc 
There were previously a dependency on system_wrappers that is now removed (uses defines to check for SEE2 instructions during compilation time instead).

Review URL: http://webrtc-codereview.appspot.com/296008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 13:03:38 +00:00
xians@webrtc.org
0dffc6449a To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. 
This patch disbale some codecs which are not considered necessary. 
Review URL: http://webrtc-codereview.appspot.com/299001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 15:35:44 +00:00
kjellander@webrtc.org
543c3eaa46 Fixing Release compilation errors 
Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 12:20:35 +00:00
andrew@webrtc.org
7eba346d30 Remove module targets due to Linux-Release errors. 
TBR=phoglund

Review URL: http://webrtc-codereview.appspot.com/277006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 19:03:38 +00:00
phoglund@webrtc.org
337dc68992 Included modules in webrtc.gyp and fixed build errors. 
Removed TODO from webrtc.gyp since it is done.

Tabs -> spaces.

Tabs -> spaces.

Tabs -> spaces.

Fixed compilation on Windows.

Added missing file.

Merge branch 'master' into fix_mac_modules

Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.

 Please enter the commit message for your changes. Lines starting

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 15:36:44 +00:00
andrew@webrtc.org
2256269a41 Enabling all common_video targets in webrtc.gyp. 
Review URL: http://webrtc-codereview.appspot.com/268001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 16:21:47 +00:00
kjellander@webrtc.org
177bb523bd Fixing system_wrappers unittests. 
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 17:10:01 +00:00
andrew@webrtc.org
3ce62fcfe4 Move merge_libs targets to their own gyp. 
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 01:03:18 +00:00
andrew@webrtc.org
19eefdc9f0 Add a unit testing framework. 
Populate it with the beginnings of a resampler unit test to have it do someting.

Also fix a bug in resampler caught with the test ;)
Review URL: http://webrtc-codereview.appspot.com/135019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-14 17:02:44 +00:00
xians@google.com
d3185fe219 refactor the gyp file to gypi file. 
Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 12:24:39 +00:00
tommi@webrtc.org
102b2270c7 First version of the peerconnection client application for Linux. 
I made several updates to the Windows version as well so that both
implementations share
a big portion of the code.
The underlying PeerConnection notifications have changed a bit since the last
update
so that there's still a known issue that I plan to fix in my next change:

  // TODO(tommi): There's a problem now with terminating connections:
  // When ending a conversation, both peers now send a signaling message
  // that indicates that their ports are closed (port=0).  The trouble this
  // causes us here is that we can interpret such a message as an invite
  // to a new conversation.  So, currently there is a bug that ending
  // a conversation can immediately start a new one.
  // To fix this I plan to change how conversations start and have a special
  // notification message via the server that prepares a client for a
  // conversation instead of automatically recognizing the first signaling
  // message as an invite.
Review URL: http://webrtc-codereview.appspot.com/112008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 15:03:52 +00:00
tommi@google.com
b0d7a87bb0 Mock implementation for the UI of the linux version of the peerconnection client. 
At this point, there's not a lot too it as it only shows what the UI will look like and basically mimics what the Windows version does presently.
Review URL: http://webrtc-codereview.appspot.com/92018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 09:03:29 +00:00
ronghuawu@google.com
35f534529b * Point the webrtc libjingle dependency to third_party_mods. 
* For unchanged files, change the third_party_mods libjingle.gyp to point to the original version of libjingle.
Review URL: http://webrtc-codereview.appspot.com/89015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 22:08:29 +00:00
niklase@google.com
f6d205aecb git-svn-id: http://webrtc.googlecode.com/svn/trunk@170 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 09:18:14 +00:00
leozwang@google.com
5e9a682f3f add command line test app to gyp build 
Review URL: http://webrtc-codereview.appspot.com/24017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-20 17:05:14 +00:00
tommi@google.com
aefcdf49ba Change the conditions for the peerconnectin_client project. 
Now the project is completely within a "win" condition which
should fix the mac build problem.
Review URL: http://webrtc-codereview.appspot.com/20021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@20 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-05-31 15:49:22 +00:00
niklase@google.com
da159d6be6 git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00