bc73871251
Remove the VPM denoiser.
...
The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 09:50:40 +00:00
eec6ecdb1e
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
...
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org , pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:09:59 +00:00
168f23faa5
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
...
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
a1bfcad3a3
Cast payload types to int for logging.
...
uint8_t gets interpreted as char and printed as such, instead of being
printed in decimal, casting them to int allows us to read what payload
types are actually used without converting them from ASCII first.
BUG=chromium:390874
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 12:33:45 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
8f1512140e
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 09:39:23 +00:00
7af12be781
Thread annotations for vie_encoder.cc/.h
...
Review URL: https://webrtc-codereview.appspot.com/8739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 14:46:31 +00:00
d11bec40b2
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 14:32:58 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
2bb1bdab8d
Preserve RTP states for restarted VideoSendStreams.
...
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00
b9f5453e29
Add boilerplate code for H.264.
...
R=mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 12:42:07 +00:00
20c1f56992
Configure RTX send status on new modules.
...
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).
As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 10:58:12 +00:00
88e0dda475
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 09:20:42 +00:00
dfdaeb92d8
Removed old code and default implementations.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 07:35:21 +00:00
1295dc6a23
Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators.
...
I think this does not introduces any contention or new deadlocks. But that is hard to verify at the moment.
BUG=chromium:388191
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6582 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-02 13:23:19 +00:00
be9d2a4549
Reserve RTP/RTCP modules in SetSSRC.
...
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 13:19:09 +00:00
a2142caa2f
Bump version number to 3.55
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 22:04:27 +00:00
1227ab89a7
GN: Add BUILD.gn files + kjellander to OWNERS
...
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.
I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.
I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.
BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
f425b55eeb
Add tests of texture frames in video_send_stream_test.
...
Also fix a bug in ViEFrameProviderBase::DeliverFrame that
a texture frame was only delivered to the first callback.
BUG=chromium:362437
TEST=Run video engine test and webrtc call on CrOS.
R=kjellander@webrtc.org , pbos@webrtc.org , stefan@webrtc.org , wuchengli@google.com
Review URL: https://webrtc-codereview.appspot.com/15789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6506 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 12:04:05 +00:00
d980307197
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
...
BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:27:19 +00:00
a1bfc50a72
Pass GYP DEPTH variable to isolate.
...
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.
Also update all our .isolate files to use the <(DEPTH)
variable.
BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.
R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 09:02:15 +00:00
cb254aac3b
Enable pacing by default and remove the option to disable it from the new API.
...
BUG=1672
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 15:12:25 +00:00
2881ab1e36
Increased kMaxRampUpDelayMs (120 to 240s).
...
Add support for triggering on encode rsd metric if its thresholds are configured. Added unit tests.
BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6410 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 08:46:46 +00:00
fbb567dacd
Add APIs to enable padding with redundant payloads.
...
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.
BUG=1812,2992
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:41:36 +00:00
734a532723
Add additional metric (relative standard deviation of encode time) for overuse detection.
...
This code is currently only for testing.
BUG=1577
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 06:35:22 +00:00
7b82c18979
Add kjellander@webrtc.org as OWNER for *.isolate
...
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.
BUG=
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 05:42:53 +00:00
b273b60154
ViEAutoTestAndroid: Unbreak compile by casting void* to jobject.
...
Sure would be nice if the try fleet used both gcc _and_ clang...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6355 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 18:59:30 +00:00
9512719569
AppRTCDemo(android): support app (UI) & capture rotation.
...
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.
BUG=2432
R=glaznev@webrtc.org , henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 18:40:44 +00:00
ef92755780
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
...
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15629005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:25:29 +00:00
e6e139159f
Android: cleanup gtest_target_type conditions.
...
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library
Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
637c55f45b
Add support of texture frames for video capturer.
...
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:00:51 +00:00
89e8ffb395
Revert "Add support of texture frames for video capturer."
...
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.
Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.
BUG=chromium:362437
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 14:12:58 +00:00
efe15355ee
Add support of texture frames for video capturer.
...
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 12:40:27 +00:00
ab6bf4f54c
Added api for getting cpu measures using a struct.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6249 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 07:43:15 +00:00
1457b4737a
First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
...
Receive stats is reset if the payload type changes. Update stats after a possible reset.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:06:04 +00:00
440e1d1053
vie_autotest_android.cc: stop referring to undefined functions.
...
The roll in r6240 exposed the fact that vie_autotest_android.cc has been
depending on vie_autotest_network.cc since forever, even though that file isn't
part of the build! #if'ing the references out to green the build.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6241 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 21:40:45 +00:00
1566ee2893
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
...
This reverts commit r6230 to re-land r6229.
ViECapturer::SwapFrame now resets timestamps.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 13:03:45 +00:00
6e98ef4b35
Fix deadlock in RegisterPreDecodeImageCallback.
...
Fixes lock-order inversion between ViEChannel::callback_cs_ and
VideoReceiver::_receiveCritSect detected on DrMemory Full which
exhibited different timing behavior.
Also removes most of the suppressions on DrMemory Full as they're able
to run again without deadlocking.
BUG=3336,3375
TEST=Run DrMemory Full trybots.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6228 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 09:41:07 +00:00
0720758f9f
Bump WebRTC version number to 3.54
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6222 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 17:51:18 +00:00
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
60015d27ae
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
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This allows use of webrtc field trials and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.
Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.
BUG=crbug/367114
R=mflodman@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 09:39:51 +00:00
54231f0662
Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.
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BUG=crbug/371714
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6166 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 23:06:23 +00:00
88abf11cad
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
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BUG=3111
TEST=try bots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 16:53:51 +00:00
a36ad6929d
Add webrtc field trials API.
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From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
4e2806d85f
Remove WEBRTC_TRACE uses in video_engine/
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Complements fixes by mflodman@.
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6136 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 08:02:22 +00:00
98c76a120d
Make vie/voe_auto_test accept non-supported flags without error.
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With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005
BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz
R=henrika@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 06:01:40 +00:00
e41dbee8a6
Reduced kMaxSampleDiffMs (limit to 22fps).
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BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6121 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 13:45:13 +00:00
46e636a3f5
Fix failing test introduced with r6111.
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Test was assuming that getting the receive estimate of a stream which hasn't received packets would return an error, new behavior is to return 0.
TBR=wu@webrtc.org
BUG=crbug/371714
Review URL: https://webrtc-codereview.appspot.com/21419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 23:17:29 +00:00
