1132 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
pbos@webrtc.org
e6dc38ea9b Lock resources in event_posix.cc.
Fixes errors reported by Helgrind from event_posix.cc when running video_engine_tests.

BUG=
TEST=helgrind,trybots
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4572 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 09:49:19 +00:00
pbos@webrtc.org
d5f4c15e8f Added missing static_cast conversion.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2061004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4568 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-19 16:35:36 +00:00
pbos@webrtc.org
e7f056ec45 Implementation and testing of PLI in new API.
BUG=2174
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2011004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4567 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-19 16:09:34 +00:00
stefan@webrtc.org
d4f607e70a Fixes to padding when driven by encoder.
- Allow padding to be sent on an ssrc which doesn't produce video, therefore
  never having the last_packet_marker_bit_ set.
- Add the random timestamp offset to all padding packets.
- Store the capture time of padding packets to properly create an offset.

BUG=2258
TEST=trybots and a new test which will be committed separately.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4566 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-19 15:55:01 +00:00
phoglund@webrtc.org
32fe90b3f9 Made all integration tests use consistent naming.
After decision by pbos@, mflodman@ et. al.

BUG=
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4565 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-19 11:40:19 +00:00
turaj@webrtc.org
f1efc57139 Implementing APIs to set maximum and minimum for latency.
cpplint warnning fixed

Ready for review

BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1971004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4563 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-16 23:44:24 +00:00
agalusza@google.com
b655985abd Added choice of decode error mode to loopback test.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4562 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-16 23:07:14 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
fischman@webrtc.org
dde7d4c6ed Roll chromium_revision 214260:217707 and gflags 45:84
gflags roll is needed mostly to pick up fixes for warnings triggered by newer
compiler/settings pulled in by the chromium roll.  Had to switch from the old
google-gflags project the current gflags project to pick up this fix (see
https://code.google.com/p/gflags/source/detail?r=74 for details).

Update android build.xml file to reflect tools moves in new SDK pulled in by the chromium_revision roll.

R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2043004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4555 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:31:30 +00:00
niklas.enbom@webrtc.org
cc9238e385 Fix OSX keydown detection. I noticed that the OSX implementation differs from Linux and Windows, and it will trigger continuously for a key that is pressed down. It would totally make sense to change this to a callback driven model, but that's a bigger change.
I need to test this before committing...

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1996004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4550 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 14:19:12 +00:00
henrike@webrtc.org
c92781737c OpenSl bug: not matching playout and record sample rate led to high or low pitch audio (depending on if playout rate was higher or lower than record rate).
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2031004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4549 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 14:13:13 +00:00
kjellander@webrtc.org
4298f73031 Revert 4547 "Isolate GYP target and .isolate files for tests"
As this breaks the FYI bots in 
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)

> Isolate GYP target and .isolate files for tests
> 
> 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/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_integrationtests
> * 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_integrationtests
> * video_capture_integrationtests
> 
> 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_integrationtests
> * vie_auto_test
> * voe_auto_test
> 
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1673004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 11:29:58 +00:00
kjellander@webrtc.org
d7a4d235d2 Isolate GYP target and .isolate files for tests
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/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_integrationtests
* 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_integrationtests
* video_capture_integrationtests

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_integrationtests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1673004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 10:02:06 +00:00
sjlee@webrtc.org
d690eab54f The video capture module for iOS.
This CL is from https://webrtc-codereview.appspot.com/1339004.

Patch this CL, then run the trunk/webrtc/build/vie-webrtc.sh.

BUG=2105
R=fischman@webrtc.org, mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1641004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4546 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-14 22:07:04 +00:00
pbos@webrtc.org
3d0019f09a Remove ViEBase::Init() call from VideoCall.
ViEBase::Init() is a no-op in the current implementation. Keeping it
there is just confusing.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4544 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-14 14:27:11 +00:00
pbos@webrtc.org
fd39e13c80 Remove VideoEngine class from new VideoEngine API.
The VideoEngine class had minimal use, so it makes more sense to bake
its functionality and config into VideoCall for a simpler API. The only
thing the VideoEngine class could do was to create VideoCalls.

BUG=2224
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4543 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-14 13:52:52 +00:00
pbos@webrtc.org
d65914360a Disable CanTransmitExtraRtpPacketsWithoutError on Windows.
Flakily crashes on Windows.

BUG=2240
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4542 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-14 09:44:19 +00:00
marpan@webrtc.org
62ecc20afb Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors".
Bot failures for Win32-Release and Linux64-Release.

TBR=pbos@webrtc.org.

Review URL: https://webrtc-codereview.appspot.com/2026004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4541 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-13 21:36:48 +00:00
pbos@webrtc.org
a05653b2c1 Disable racy part of RunsRtpRtcpTestWithoutErrors.
Disabled part as suggested in bug 1790, but without breaking it up into
multiple tests. These tests will be made redundant by tests for the new
API, and it would take far too long to clean these up properly.

BUG=1790
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2022004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4539 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-13 14:27:20 +00:00
wuchengli@chromium.org
e1051b0731 Add native_handle.h to gyp.
BUG=http://crbug.com/170345
TEST=Build all.
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4538 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-13 05:53:38 +00:00
minyue@webrtc.org
db1cefc14e To allow the propagation of under-run in NetEq.
BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1974004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4537 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-13 01:39:21 +00:00