90 Commits

Author SHA1 Message Date
fischman@webrtc.org
f792d17870 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/.
(needs to land separately from the rest because PRESUBMIT)

Original review URL: https://webrtc-codereview.appspot.com/9229004

BUG=2168
TESTED=trybots
RISK=P3 (code is unused ATM)

Patch from Sajid Hussain <shussain@temasys.com.sg>.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5671 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 17:12:08 +00:00
fischman@webrtc.org
7bd4a27502 VideoCaptureAndroid: don't deliver frames after stopCapture().
Because stopCapture() and onPreviewFrame() are called on different threads, and
are both synchronized, it's possible for onPreviewFrame() to commence execution
after stopCapture() has completed, causing a SEGV because the native code is no
longer prepared to accept frames.
Clarify the contract around synchronized methods in this class to hopefully
avoid similar bugs in future.

BUG=2947
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5639 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 18:17:55 +00:00
fischman@webrtc.org
8685af7ea0 Remove "Too long processing time of Incoming frame" logspam.
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.

BUG=2732
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
mallinath@webrtc.org
7433a088d2 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.

TBR=andresp@webrtc.org

> Revert 5421 "Fix deadlock on register/unregister observer while ..."
> 
> Failure to compile on Chromium Internal bots, because of API changes.
> 
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
> 
> You need to follow the steps mentioned in 
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
> 
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
> 
> > Fix deadlock on register/unregister observer while there is a an going callback.
> > 
> > BUG=2835
> > R=mallinath@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/7119005
> 
> TBR=andresp@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7679004

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 00:56:02 +00:00
fischman@webrtc.org
932b0193e7 VideoCaptureAndroid: stop preview in opposite order of starting.
While the SDK documentation doesn't prescribe a required shutdown order, good
hygiene suggests stopping should happen in reverse order of starting.  It also
seems to relieve a crash in the system capturer on at least the Galaxy Note 10.

BUG=2793
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:32:05 +00:00
mallinath@webrtc.org
18586d38bc Revert 5421 "Fix deadlock on register/unregister observer while ..."
Failure to compile on Chromium Internal bots, because of API changes.

http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio

You need to follow the steps mentioned in 
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.

Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.

> Fix deadlock on register/unregister observer while there is a an going callback.
> 
> BUG=2835
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7119005

TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:00:57 +00:00
andresp@webrtc.org
8d375c95b7 Fix deadlock on register/unregister observer while there is a an going callback.
BUG=2835
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 23:09:25 +00:00
kjellander@webrtc.org
f9bdbe3619 Roll chromium_revision 232627:238260
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
pbos@webrtc.org
2ffb149c2c Replace VideoFrameI420 with I420VideoFrame.
Gives one less struct/class for I420 video frames.

BUG=2657
R=mflodman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5160 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 13:10:13 +00:00
sheu@chromium.org
5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.

TBR=niklas.emblom@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/3269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 23:41:04 +00:00
sheu@chromium.org
74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:48:16 +00:00
sheu@chromium.org
d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.

Revert while build breakage is fixed.

BUG=None
TBR=niklas.emblom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 21:20:15 +00:00
sheu@chromium.org
1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 20:32:28 +00:00
pbos@webrtc.org
e05362916c Make sure the first frame isn't dropped.
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 09:02:30 +00:00
fischman@webrtc.org
4598380860 Android: enable camera video stabilization when available.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2347005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4929 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 20:14:19 +00:00
kjellander@webrtc.org
3f9288f987 Add APK and isolate target for video_engine_tests
Add .isolate file and _run target for video_engine_tests.

Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844)

Update modules_unittests.isolate with new NetEq4 reference files
needed.

TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 18:20:38 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
kjellander@webrtc.org
2a97317953 Fix include of isolate.gypi
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.

The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.

TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).

I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).

I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.

Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc

BUG=1916
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 19:31:16 +00:00
pbos@webrtc.org
1c974ef5e3 Remove include_dirs from video_capture.
BUG=1662
TEST=compile on trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2303005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4880 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-30 15:32:10 +00:00
fischman@webrtc.org
69fc315fd9 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector.
The map was just mapping an index to a pointer of a POD, so the code is easily
simplified by using a vector (with implicit index key) and the POD as a value.
(also fixes a leak in the windows code, which lacked a virtual dtor for
VideoCaptureCapabilityWindows but was deleting through a base pointer).

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2298004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4840 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 17:01:42 +00:00
fischman@webrtc.org
334865e2a1 Re-enable VideoCaptureTest.CreateDelete
Previously the test insisted on non-zero delay, but 0 is not a crazy delay value
(esp. on a fake camera device!).  Instead we now test for delay>=0 being set at
all.

BUG=2405
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4813 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 17:58:45 +00:00
stefan@webrtc.org
9c74be7bd1 Disable flaky video capture test.
BUG=2405
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4801 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-20 08:52:31 +00:00
sjlee@webrtc.org
e6ac163145 This is related to https://code.google.com/p/webrtc/issues/detail?id=846
BUG=846
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2224004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4760 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 05:15:19 +00:00
mallinath@webrtc.org
82a846f0cb Adding Ami to the video renderer and capturer modules.
TBR=fischman@webrtc.org,wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2202006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4735 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:43:17 +00:00
andresp@webrtc.org
77bf5c28c8 Clean capture timestamp code.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2134004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4675 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:35:43 +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
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
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
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
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
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
4ca7d3f9fe Replace MapWrapper with std::map<>.
MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 19:51:57 +00:00
pbos@webrtc.org
a3b7406219 Remove unused unreferenced code in webrtc/
The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.

BUG=
R=andrew@webrtc.org, henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, turaj@webrtc.org, wu@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1945004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:47:51 +00:00
fischman@webrtc.org
d6134c7cfd PeerConnectionTest.java: make the test work for the bots' v4l2loopback.
- Make the test agnostic to the actual resolution used, since v4l2_file_player
  is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
  v4l2_file_player is feeding.

Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.

BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1891004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 20:43:15 +00:00
sergeyu@chromium.org
099b8c9e8e Update include paths in device_info_external.cc
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1875004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4401 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 18:41:43 +00:00
niklas.enbom@webrtc.org
8e3bbedacd Fix include path in video_capture_external.cc
Fix build error introduced in r4337

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1873004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4397 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 16:55:58 +00:00
fischman@webrtc.org
c6d5b50b41 AppRTCDemo: build fixes for iOS build in webrtc
BUG=1421,1450,1451
TESTED=git try, also the same patch (along with a bunch of other, non-webrtc changes) in a libjingle checkout allows building iOS AppRTCDemo
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4371 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-18 02:02:07 +00:00
pbos@webrtc.org
df119c9a45 Remove dead video_capture for QuickTime.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4339 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 18:08:13 +00:00
pbos@webrtc.org
a9b74ad716 Include files from webrtc/.. paths in video_capture/.
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4337 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 10:03:52 +00:00
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:31:18 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
pbos@webrtc.org
504af45a6f Diff NTP and internal once in VideoCaptureImpl.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1754004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-02 10:15:43 +00:00
fischman@webrtc.org
546c91dc2e Build all java files into jar for each module on Android
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1696004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-01 17:52:39 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests.
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 20:09:44 +00:00
fischman@webrtc.org
dd97ef4e28 Revert 4211 "Build all java files into jar for each module on An..."
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files

> Build all java files into jar for each module on Android
>
> BUG=
> R=fischman@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1636004
>
> Patch from Jeremy Mao <yujie.mao@intel.com>.

TBR=fischman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1660005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 17:39:29 +00:00
fischman@webrtc.org
1374965680 Build all java files into jar for each module on Android
BUG=
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1636004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4211 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 23:34:27 +00:00
mflodman@webrtc.org
3ba883f0fc Removing functionality for inserting pre-encoded frames instead of raw
video frames. The functionality hasn't been used for a long time and
should be done properly if used in the future.

This is a pre-step for implementing CPU overload control.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1630004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 13:57:57 +00:00
fischman@webrtc.org
8d6eb56085 Avoid NPE crash on Android platforms that don't support getting preview framerate.
- catch Camera.setParameters() signaling errors through RuntimeException (!)
- make video_demo_apk rebuild when .java sources change

BUG=1778
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1493004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4059 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 17:33:31 +00:00