213 Commits

Author SHA1 Message Date
pbos@webrtc.org
841c8a44bb Rename VideoCall to Call.
Call should encompass more than video, there's no point in calling it
VideoCall.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 15:04:25 +00:00
pbos@webrtc.org
0181b5f8dd ExternalVideoDecoder for new VideoEngine API.
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.

BUG=2346,2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2172004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 08:26:30 +00:00
fischman@webrtc.org
c7f708679d Clamp camera id to legal values.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4694 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 18:17:45 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +00:00
pbos@webrtc.org
95e51f509c Remove send and receive streams when destroyed.
Fixes crash where packets were sent to a receive stream that had been
destroyed but not removed from the ssrc mapping from call to receiver.
Added a repro case that reliably crashed before the fix.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2161007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4681 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:38:54 +00:00
pbos@webrtc.org
7e1bf318bf Allow unknown flags in test_main.cc.
Adds AllowCommandLineParsing to allow us to ignore "--no-sandbox" given
by new TSanV2 bots. Not ignoring this flag prevents the test from
running on this machine. Also removing unnecessary asserts that clutter
code.

BUG=
TEST=Locally running video_engine_tests with --no-sandbox.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4679 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 10:27:46 +00:00
mflodman@webrtc.org
e2d4da6586 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter.
BUG=2346
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4677 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 14:21:57 +00:00
mflodman@webrtc.org
06f1f74331 Disable EngineTest.ReceivesPliAndRecoversWithNack.
The test times out on Linux memcheck bot at times.

BUG=2348

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4674 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:00:07 +00:00
pbos@webrtc.org
cb5118c14c Add FakeEncoder to VideoSendStream tests.
Breaks out config part of FakeEncoder from VideoSendStream tests to
FakeEncoder. Also sets FakeEncoder as encoder for VideoSendStream tests.
Anticipated speedup didn't happen as VP8 is still initialized by default
when creating channels in the old API. This will be sped up when moving
off the old API as VP8 won't be enabled by default.

BUG=2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2155004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4659 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 09:10:37 +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
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
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
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
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
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
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
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
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
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
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
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
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
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
pbos@webrtc.org
468e19aa93 Signal when shutting down DirectTransport.
Avoids starting the network thread when there are no packets to be read.
This allows the transport to shut down directly, which makes tests using
it able to quit faster, and not have to wait up to 10ms.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4524 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 14:28:00 +00:00
pbos@webrtc.org
9668467d87 Run loopback tests with network thread.
Running with a network thread provides a more realistic simulation. Like
a real network, packets are handed off to a socket, or buffer, and then
the call returns. This prevents weird scenarios when both the sending
side and receiving side are on the call stack simultaneously, which can
cause deadlocks as locks could otherwise be taken simultaneously in both
the sender and receiver order by the same thread.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4522 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 12:59:04 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769.
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 20:36:57 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +00:00
pbos@webrtc.org
4052370e89 Use RtpHeaderParser in VideoCall implementation.
BUG=1827
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1962004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4483 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 12:49:22 +00:00
pbos@webrtc.org
bbb07e69e5 Glue code and tests for NACK in new VideoEngine API.
The test works by randomly dropping small bursts of packets until enough
NACKs have been sent back by the receiver. Retransmitted packets are
never dropped in order to assure that all packets are eventually
delivered. When enough NACK packets have been received and all dropped
packets retransmitted, the test waits for the receiving side to send a
number of RTCP packets without NACK lists to assure that the receiving
side stops sending NACKs once packets have been retransmitted.

BUG=2043
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1934004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4482 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 12:01:36 +00:00
pbos@webrtc.org
7fb9ce0cf5 Fix send times in video_full_stack.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1947004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4481 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 09:29:50 +00:00
fischman@webrtc.org
d3ae3c7b1f Unbreak clang/android build of webrtc.
TESTED=All target builds once more with clang=1.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4460 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-31 23:53:07 +00:00
mflodman@webrtc.org
6879c8adad Hooking up first simple CPU adaptation version.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1767004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 11:35:00 +00:00
yujie.mao@webrtc.org
129afc29fb Correctly rebuild WebRTCDemo after jni/ source file changes
BUG=1980
TEST=Modify source file under jni/ and WebRTCDemo will rebuild
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1831004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4377 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-20 04:43:08 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
pbos@webrtc.org
69215d8432 Include files from webrtc/.. paths in video_engine/.
BUG=1662
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1759005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 15:02:02 +00:00
pbos@webrtc.org
adf23a55f8 Direct3D renderer for new VideoEngine API tests.
TEST=Rendered video in video_loopback test.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1573004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 14:07:56 +00:00
pbos@webrtc.org
f3f1358360 Fixed implicit-int-conversion bugs.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 14:04:46 +00:00
stefan@webrtc.org
cab716cc7d Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
TBR=henrikg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1776005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 13:43:24 +00:00