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Author SHA1 Message Date
pbos@webrtc.org
5860de02aa Implement NACK over RTX for VideoSendStream. 
BUG=2231
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2197008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 13:01:47 +00:00
pbos@webrtc.org
5c678eabd9 Implement 'abs-send-time' extension in VideoSendStream. 
BUG=2229
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 19:00:39 +00:00
pbos@webrtc.org
2902328cce Implement 'toffset' extension in VideoSendStream. 
BUG=2229
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 10:14:56 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
pbos@webrtc.org
df531a2eee Test that VideoSendStream responds to NACK. 
BUG=2228
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 14:56:33 +00:00
pbos@webrtc.org
744fbc7fe4 Split up EngineTests and RampupTests. 
This allows having one group of tests per file, the test files are
long enough as they are.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 09:26:25 +00:00
elham@webrtc.org
a19c9f4173 Updated WebRTC version to 3.41 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:23:44 +00:00
pbos@webrtc.org
7ebf0e7f44 Remove include_dirs from video_engine_core.gypi. 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 16:56:31 +00:00
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
b2c8a952a7 Improving padding rules and breaking out bw allocation to ViEEncoder. 
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:58:01 +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
mikhal@webrtc.org
f1e807c0e5 Removing FrameForStorage 
R=pwestin@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 22:34:41 +00:00
andrew@webrtc.org
9080518a39 Restore severity precondition to logging.h. 
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.

Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort

Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled:                          666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled:       673 ms (1.01x)

BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2160005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 16:40:43 +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
mflodman@webrtc.org
8d32066073 Changed method name. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:30 +00:00
mflodman@webrtc.org
814d5e9133 Renamed method. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:18 +00:00
mflodman@webrtc.org
d51bcffc1e Function name change. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:09 +00:00
mflodman@webrtc.org
dfbf52baac Fixing capture frame race in ViECapturer. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:44:57 +00:00
pbos@webrtc.org
a957570d62 Overuse detection based on capture-input jitter. 
This is believed to be more reliable in real-world cases. The camera seems to fall behind sooner than the encoder starts taking too long time encoding, so this is believed to be an earlier trigger.

BUG=2325
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 17:16:32 +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
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
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
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
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
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