Commit Graph

6358 Commits

Author SHA1 Message Date
aluebs@webrtc.org
021e76fd39 Add support for WAV output in audioproc
The default output is a WAV file, except if the --pcm_output flag is set.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 18:12:00 +00:00
jiayl@webrtc.org
52055a276d Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:12:25 +00:00
brettw@chromium.org
afa77cd803 Add direct_dependent_config to desktop_capture in GN build.
This allows us to remove some configs in the Chrome build that should come
automatically from depending on this target.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7067 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:00:55 +00:00
pbos@webrtc.org
ceb956b29d Abort Negotiate() if DoCreateOffer() fails.
Addressing crash in test.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/19239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 15:27:49 +00:00
kjellander@webrtc.org
d57c95fde4 Change Chromium .gclient to not use Managed mode.
Since the sync_chromium.py script always passes --revision
to the gclient sync command, we don't need to have
managed=True in the .gclient file.
This will avoid a warning that confuses our developers.

BUG=3776
TESTED=Removed my chromium/.last_sync_chromium and performed
a gclient sync with this patch applied. No warning complaining
about Managed mode appears.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7065 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:58:55 +00:00
andresp@webrtc.org
fa822b940f Fix strange owners files with comments that crashs "git cl presubmit"
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7064 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:25:07 +00:00
kjellander@webrtc.org
79ee97cf43 [MIPS] Fix gn gen failure for MIPS in webrtc
Fixes the following failure for mips:
"ERROR at //third_party/webrtc/BUILD.gn:136:7: Undefined variable for +=.
      cflags += [ "-mhard-float" ]
      ^-----
I don't have something with this name in scope now."

BUG=3441
TEST=In Chromium. Passing compile locally on Linux using:
gn gen out-gn/mips --args="is_debug=false os=\"android\" cpu_arch=\"mipsel\"" --verbose &&  ninja -C out-gn/mips all
gn gen out-gn/arm --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" --verbose &&  ninja -C out-gn/arm all
gn gen out-gn/x86-linux --args="is_debug=false os=\"linux\"" --verbose &&  ninja -C out-gn/x86-linux webrtc

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15349004

Patch from Gordana Cmiljanovic <Gordana.Cmiljanovic@imgtec.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7063 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:10:49 +00:00
houssainy@google.com
38ef664418 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between
/borwser and /android

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:44:47 +00:00
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files.
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
bjornv@webrtc.org
3cbd6c26c8 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition.
This changes some method signatures to better reflect how callers are actually
using them.  This also has the tendency to make signatures more consistent about
e.g. using int (instead of int16_t) for lengths of things like vectors, and
using int16_t (instead of int) for e.g. counts of bits in a value.

This also removes a couple of functions that were only called in unittests.

BUG=3353,chromium:81439
TEST=none
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:21:44 +00:00
henrik.lundin@webrtc.org
f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo
The name of the copy method was confusing. This change makes the
code easier to read where the method is used.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 10:58:43 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths.
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 09:55:40 +00:00
kwiberg@webrtc.org
51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_
R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 08:42:44 +00:00
henrik.lundin@webrtc.org
7825b1abf9 Add support for multi-channel DTMF tone generation
This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:39:21 +00:00
pbos@webrtc.org
bcb6bcfe6c Remove HybridVideoEngine.
This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:32:26 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:07:44 +00:00
stefan@webrtc.org
01581da711 Fix audio/video sync when FEC is enabled.
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.

BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 06:48:14 +00:00
andresp@webrtc.org
bfd7a8c448 Fix compile errors on webrtc/base.
R=fbarchard@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7052 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 04:59:52 +00:00
andresp@webrtc.org
0229cbae33 Remove ambiguous call to MakeCheckOpString.
BUG=3777
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7051 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 04:53:29 +00:00
thorcarpenter@google.com
95c2458766 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files.
"gcl try" fails to upload these large files so adding them independently.

R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 23:17:36 +00:00
fbarchard@google.com
9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 23:05:07 +00:00
tkchin@webrtc.org
5b83af49c1 Fix leak of NSAutoreleasePool.
This looks like something that's no longer applicable. From what I saw this code path isn't on a static initializer that runs before main. Should be okay to drain (release) pool outside of this scope.

BUG=3659
R=henrike@webrtc.org, jiayl@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7048 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 22:53:34 +00:00
buildbot@webrtc.org
609f987488 (Auto)update libjingle 74696326-> 74723281
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 21:50:32 +00:00
henrike@webrtc.org
1b8b4c4959 Revert 7041 " Audio codecs to include webrtc/typedefs.h"
Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio

R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/19219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 19:42:16 +00:00
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 16:49:04 +00:00
pbos@webrtc.org
26c0c41a06 Network up/down signaling in Call.
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 16:17:12 +00:00
pbos@webrtc.org
ebee401230 Remove flake in SendsLowerResolutionOnSmallerFrames.
Speculative fix for break on Linux64 Release. It looks like the second
frame is being dropped which is likely because the two frames are sent
too close to eachother. Adding a delay of 33ms in between them to make
sure the second one isn't dropped.

R=minyue@webrtc.org
TBR=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7043 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 15:52:02 +00:00
pbos@webrtc.org
c4175b9fdf Set resolution based on incoming VideoFrames.
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/17269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 15:25:49 +00:00
andresp@webrtc.org
9730d3aae9 Audio codecs to include webrtc/typedefs.h
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 14:37:18 +00:00
kjellander@webrtc.org
0372b93118 Partial revert of r7014 (Android APK refactor)
This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 14:34:46 +00:00
aluebs@webrtc.org
bac072667b Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes
The sizes saved in the aecdumps were always the input length, and this is not necessarily true when there is a change in sample rate. But the sample rates dumped are correct, so we can calculate the sizes from them knowing that we use 10ms chunks.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7039 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 13:39:01 +00:00
minyue@webrtc.org
adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 12:28:06 +00:00
stefan@webrtc.org
0a214ffa8a Setting marker bit on DTMF correctly
BUG=1157
R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 11:46:54 +00:00
aluebs@webrtc.org
74cf916924 Fix issues in audioproc for float aecdumps
* The right buffer size is used to dump to file when the output sample rate is different from the input one.
* The percentage of processed chunks is calculated correctly when float data available.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7036 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 11:05:01 +00:00
bjornv@webrtc.org
48f2568d89 audio_processing/nsx: Bug fix that could cause divide by zero
In the fixed point version of the Noise Suppression. At one place we subtract a value in the wrong Q-domain, which later may cause a divide by zero. Going through the floating point code that particular variable should be zero if this happens, which is what the old code tried to accomplish, but in an awkward way.

The bug has been there since development, so the likelihood of actually get a divide by zero is very small.

BUG=chromium:407812
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7035 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 07:58:37 +00:00
minyue@webrtc.org
d944a6887d Suppressing VideoAdapterTest.AdaptResolutionWide and VideoAdapterTest.AdaptResolutionNarrow on DrMemory
BUG=webrtc:3771
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7034 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 07:43:32 +00:00
buildbot@webrtc.org
72e448559d (Auto)update libjingle 74628537-> 74648573
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7033 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 00:43:48 +00:00
tkchin@webrtc.org
90750482fa Remove deprecated RTCVideoRenderer constructor.
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track.

BUG=3341
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 20:50:00 +00:00
andrew@webrtc.org
34a6764981 Remove the checks.h dependence on logging.h in a standalone build.
logging.h apparently drags in a lot of undesirable dependencies. It was
only required for the trivial LogMessageVoidify; simply add an
identical FatalMessageVoidify instead.

Keep the include in a Chromium build to still have the override
mechanism use Chromium's macros.

Bonus: Add the missing DCHECK_GT (noticed by bercic).

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7031 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 19:00:45 +00:00
stefan@webrtc.org
8e24d87778 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking.
BUG=3681
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7030 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 18:58:24 +00:00
pbos@webrtc.org
9f341283f6 Remove WebRtcVideoEngine::default_codec_format().
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7029 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 16:33:09 +00:00
pbos@webrtc.org
03655143db Remove files from talk/PRESUBMIT.py.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7028 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 16:17:36 +00:00
henrike@webrtc.org
d72a7599d4 Create a copy of talk/xmllite under webrtc/xmllite.
BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7027 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 15:41:12 +00:00
kjellander@webrtc.org
6f729e8a74 Disable video_engine_tests and webrtc_perf_tests on Android.
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 15:13:55 +00:00
henrik.lundin@webrtc.org
ee0fb187a5 Divide-by-zero problem in NetEq's Normal::Process fixed
Adding a couple of tests that tries to trigger a certain divide-by-zero
issue. The tests triggered the issue, but this CL also includes a fix
for this.

BUG=3761
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7025 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 13:22:11 +00:00
kjellander@webrtc.org
94da2034b0 Remove retired android_apk[_rel] trybots from PRESUBMIT.py
BUG=webrtc:3741
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7024 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 13:05:58 +00:00
kjellander@webrtc.org
324b72dda6 Disable video_capture_tests for Android.
BUG=3768
TESTED=Passing the steps in webrtc:3768
TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7023 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 12:37:50 +00:00
kjellander@webrtc.org
e281f7fba3 GN: Update webrtc/base to recent GYP changes.
Update the webrtc/base/BUILD.gn file to reflect
webrtc/base/base.gyp changes between r6438 and r7011.

BUG=3441
TESTED= Trybots + compilation with a standalone WebRTC checkout:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default

Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 11:22:06 +00:00
andresp@webrtc.org
468516c959 RTCBot is a framework that allows to write tests where logic runs on a single
host that controls multiple endpoints ("bots"). Thus allowing to create more
complex scenarios that would otherwise require non-trival signalling between
multiple parties.

R=houssainy@google.com, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7021 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 10:52:54 +00:00
kjellander@webrtc.org
561a9eccc5 Update checkedeps.py rules in DEPS.
Add allow-rules as well in addition to the
disallow-rule in r7014.

BUG=
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7020 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 09:39:35 +00:00