Commit Graph

4063 Commits

Author SHA1 Message Date
phoglund@webrtc.org
02421fcf0b Corrected documentation on webrtc_test.sh.
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2122004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4628 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 14:00:10 +00:00
kjellander@webrtc.org
e141373b8a Add isolate configuration for Android for all tests.
In https://code.google.com/p/webrtc/source/detail?r=4407
henrike@ added the path to the WebRTC resources and
data directories for Android that are required in order to
use isolate for test execution on Android.

This CL adds similar entries to the rest of the .isolate
files added in
https://code.google.com/p/webrtc/source/detail?r=4590.

It also removes three accidentally added .isolate files that originated
from old test names:
* audio_device_test_api
* video_capture_module_test
* video_render_module_test

BUG=1882,1916
TEST=trybots passing.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 12:10:09 +00:00
tina.legrand@webrtc.org
89502c1ca8 Memory and tsan tests: Turning off renamned tests
In r4625 AudioCodingModuleTest.RunAllTests was broken down to a number of
smaller tests. This CL turns off these tests, as RunAllTests was turned off
before.

TBR=kjellander@webrtc.org

BUG=issue2173

Review URL: https://webrtc-codereview.appspot.com/2121004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4626 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 09:38:10 +00:00
tina.legrand@webrtc.org
ee92b664b3 Re-organizing ACM tests
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.

While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.

I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().

BUG=issue2173
R=minyue@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1961004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 07:33:51 +00:00
elham@webrtc.org
d6fef9d380 Fixing SetDecodeErrorMode build error
- got introduced when reverting r4562

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2118004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4624 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 23:59:38 +00:00
elham@webrtc.org
814e28413d Revert r4562
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2117004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4623 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 23:21:03 +00:00
sergeyu@chromium.org
01cb3ad883 Fix image flipping for OpenGL-based screen capturer on Mac.
I broke captured image flipping when refactoring this code while it was
still in chromium. Previously we had CaptureData that was returned from
capturers with correctly inverted stride, but frames were still stored
with positive stride. CaptureData was removed and so the returned frames
always had positive stride, which is not correct. Now ScreenCapturerMac
uses frames with inverted stride when capturing using OpenGL.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2105004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4621 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 21:48:56 +00:00
fischman@webrtc.org
e3de6b1e90 Enable ObjC build by default and reenable 64-bit mac libjingle build
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2080004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 19:31:21 +00:00
fischman@webrtc.org
4498d013f6 apprtc: rationalize whitespace
- Remove ^M DOS line endings
- Remove trailing whitespace
- Remove leading 2-space indents from files that have carried this indent since   their contents was removed from within enclosing contexts that required it.
- Add a newline to avoid 82-column line.

R=vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2112004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4619 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 18:01:28 +00:00
fischman@webrtc.org
5a035b4279 apprtc: add ctrl+i Info window showing gathered ICE candidate types
R=vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4617 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:44:38 +00:00
elham@webrtc.org
6dc45a67ee Updated WebRTC version to 3.40
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2111004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4616 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:30:54 +00:00
mallinath@webrtc.org
af84d782f0 Initialize ssl_role_ to the default role in FakeTransportChannel
constructor.
This is needed as BaseSession tests can query the transport channel
without creating dtlstransportchannel ( as they are unaware of the
underlying implementation).

This will also fix the memcheck error in webrtc bots.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2110004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4615 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:14:13 +00:00
mikhal@webrtc.org
f31a47abdc VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that,
BUG=
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2077004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4614 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:10:11 +00:00
phoglund@webrtc.org
c9fa0fede5 Removed build status tracking, refreshed front page.
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2106004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4613 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 08:45:22 +00:00
sergeyu@chromium.org
f1fd9d0c5c Fix compilation on windows after libjingle updated.
For some reason MSVC doesn't use implicit char[]->std::string 
conversion when comparing char[] and std::string in EXPECT_EQ.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4611 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-24 01:02:36 +00:00
sergeyu@chromium.org
492e315400 Update gyp file after libjingle roll.
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2103004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4609 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-24 00:06:43 +00:00
sergeyu@chromium.org
0be6aa0665 Update talk to 51314459
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4608 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 23:21:25 +00:00
mikhal@webrtc.org
b2c28c3699 Relanding 4597 - Don't force key frame when decoding with errors.
Makes sure that incomplete key frame or delta frames will be released from the JB when decoding with errors.
The decoder in turn will trigger a PLI until a complete key frame is received in order to start a session.

TBR=stefan@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/2097004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4607 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 21:54:50 +00:00
henrike@webrtc.org
442709ea96 Disable broken test and add suppressions.
BUG=2299,1205

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2101004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4606 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 21:42:01 +00:00
sergeyu@chromium.org
9f282403f2 WindowCapturer implementation for Linux.
Window enumeration is based on the code used by hangouts plugin
(see libjingle/talk/base/linuxwindowpicker.cc). XServerPixelBuffer
is used to capture windows. It had to be refactored to support window
capturing (previously it worked only for the whole screen).

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1741004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4605 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 18:22:12 +00:00
henrike@webrtc.org
563910bde3 Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests).
TBR=wu@webrtc.org

BUG=2296

Review URL: https://webrtc-codereview.appspot.com/2098004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4604 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 16:16:03 +00:00
henrike@webrtc.org
69a774fc42 Suppresses another tsan warning. Warning is reported here: http://chromegw/i/client.webrtc/builders/Linux%20Tsan/builds/460/steps/memory%20test%3A%20libjingle_peerconnection_unittest/logs/D5CAED6268DAACB7
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2096004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4603 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 14:38:15 +00:00
henrike@webrtc.org
c0b1a280ab Some tests were not disabled correctly as it should be DISABLED_* not DISABLE_*.
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4602 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 14:32:21 +00:00
pbos@webrtc.org
74fa4893f9 Remove newapi:: namespace for typenames without overlap.
Typing newapi:: everywhere is very verbose, and doesn't add any real
value. The new API is still separated from other code by being in
separate directories, such as internal/ or new_include.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2075004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4601 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 09:19:30 +00:00
henrike@webrtc.org
ceea41d135 Revert 4597 "Don't force key frame when decoding with errors"
> Don't force key frame when decoding with errors
> 
> BUG=2241
> R=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2036004

TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2093004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4600 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 00:53:24 +00:00
sergeyu@chromium.org
eef29ec6cf Implement window capturer for OS X.
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2055005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4599 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 00:39:46 +00:00
fischman@webrtc.org
d26f791273 AppRTCDemo(android): allow audio-only calls to test iOS interop
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2091004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4598 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 23:50:48 +00:00
mikhal@webrtc.org
44af55cc44 Don't force key frame when decoding with errors
BUG=2241
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2036004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4597 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 23:29:43 +00:00
henrike@webrtc.org
61b262c427 Disable tests according to issues: 1205,2272,2288,2290,2291
BUG=1205,2272,2288,2290,2291
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4596 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 20:27:49 +00:00
henrike@webrtc.org
7666db79fa Update talk to 51242664.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2090005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 14:45:42 +00:00
pbos@webrtc.org
c095f510b6 Remove template usage of typeless enum in fake_encoder.
Removes clang warning preventing compile.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2087005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4593 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 12:34:58 +00:00
pbos@webrtc.org
013d994583 Enabling and testing RTCP CNAME in new API.
BUG=2232
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2076004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4592 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 09:42:17 +00:00
stefan@webrtc.org
360e376872 Adds two tests for verifying padding and ramp-up behavior.
BUG=1837
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2073004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4591 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 09:29:56 +00:00
kjellander@webrtc.org
3365422c41 Isolate GYP target and .isolate files for tests
This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.

Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/googletest/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_tests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_tests
* video_capture_tests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_tests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above. WebRTC trybots passing. Created a Chromium checkout with third_party/webrtc ToT and this patch applied, passing the runhooks step.
BUG=1916
R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2056004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 07:57:00 +00:00
braveyao@webrtc.org
c028ee2bf2 Android audio opensles: random deadlock in stopRecording().
BUG=2201
Test=WebRTCDemo

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4589 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 03:14:34 +00:00
stefan@webrtc.org
286fe0b04d Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
...and fixes the RTCP bug.

BUG=2277
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:58:21 +00:00
mikhal@webrtc.org
dbf6a81cb5 Follow-up changes to kSelectiveErrors
Committing cl for agalusza (cl 1992004)
TEST = trybots
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2085004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4587 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:40:47 +00:00
henrike@webrtc.org
60bdb07a16 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots.
BUG=2277,2278
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4586 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 19:55:53 +00:00
henrike@webrtc.org
a0218a84d1 Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
> Reverts a second set of reverts caused by a bug in a dependency.
> 
> Revert "Revert r4328"
> 
> Revert "Revert r4322 "Support sending multiple report blocks and keeping track
> of statistics on"
> 
> BUG=1811
> R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2072004

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2087004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 19:44:13 +00:00
hta@webrtc.org
cc39484770 IP address display from stats.
This CL demonstrates a couple of methods to extract more complex properties from the stats that are linked via stats IDs.

RISK=P3
TESTED=manual test
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1667005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4584 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 17:00:54 +00:00
phoglund@webrtc.org
17018ed39c Added perf summary pages to the dashboard server.
The purpose is to make the WebRTC performance metrics easier to overview.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2081004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4583 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 16:31:12 +00:00
stefan@webrtc.org
1a65d6c36b Reverts a second set of reverts caused by a bug in a dependency.
Revert "Revert r4328"

Revert "Revert r4322 "Support sending multiple report blocks and keeping track
of statistics on"

BUG=1811
R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2072004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 16:22:21 +00:00
pbos@webrtc.org
fbf0f69bf8 Call SetExecutablePath from test_main.cc
Fixes crash in video_engine_tests on bots, that were unabled to locate
the resource file.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2083004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4581 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 16:00:15 +00:00
pbos@webrtc.org
4c96601aed Make FrameGeneratorCapturer own frame_generator.
Fixes memleaks where test::FrameGenerator::Create() was used to create
frame_generator, but it was never freed. Since the frame generator
shouldn't be used concurrently it's easiest if FrameGeneratorCapturer
take ownership of the instance.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2047005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4580 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 12:07:37 +00:00
phoglund@webrtc.org
abc1ed37c6 Merging video_full_stack_tests and video_engine_tests.
The reason is that we want to have as few test targets as possible to simplify bot configuration. It's also more convenient for developers since it will be trivial to introduce more perfing tests.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2068004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4579 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 12:06:03 +00:00
fischman@webrtc.org
d0f4c2185b iOS: unbreak the build following r4546
BUG=2255
R=niklas.enbom@webrtc.org, sjlee@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2078004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4577 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 22:16:55 +00:00
wu@webrtc.org
ebe68aad44 Fix memory leak in portallocatorsessionproxy_unittest.
Remove the suppressions that have been fixed.

BUG=1972,2263
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2062005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4576 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 21:14:39 +00:00
kjellander@webrtc.org
cbdb9d1c69 Add comment about updating webrtc.DEPS when rolling gflags
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2070004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4575 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 16:18:35 +00:00
kjellander@webrtc.org
25b39ab1a6 Document updating gflags and remove code duplication.
When rolling the google-gflags dependency, there might be
a need of updating the generated configuration files. I added
a instructions to the README.webrtc file for doing that.

This CL also removes duplicated configuration headers so we
only separete the ones that differs (Windows and everything
else).

BUG=2251
TEST=none
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2046004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4574 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 16:17:10 +00:00
pbos@webrtc.org
119a1ccdca VideoSendStream SSRC test.
Verifies that the VideoSendStream starts sending the set SSRC over RTP.

BUG=2227
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4573 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 13:14:07 +00:00