Commit Graph

6698 Commits

Author SHA1 Message Date
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
kjellander@webrtc.org
76a42577ad Remove build_with_chromium==1 conditions for Android
Most of these changes were done in r7014, but a few targets
were missed. This should make these tests run better
(but they might still be failing due to webrtc:3764).

BUG=webrtc:3741
TESTED=Local compilation using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7019 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 08:40:39 +00:00
aluebs@webrtc.org
841f58f64c Unpacking aecdumps generates wav files
BUG=webrtc:3359
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 07:51:51 +00:00
kjellander@webrtc.org
c3f42f37b5 Fix audio_decoder_unittests.isolate
In r6427 all .isolate files except
audio_decooder_unittests.isolate was updated to use the
<(DEPTH) variable instead of relative paths.
This started breaking the Android bots after committing
r7014.

BUG=3741
TBR=phoglund@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/23409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 15:06:14 +00:00
henrik.lundin@webrtc.org
8dbeb5b301 Adding more codecs to the AcmSenderBitExactness
New tests include iSAC-swb, PCM16b (8, 16, 32 kHz; mono and stereo),
PCM A/u (mono and stereo), iLBC, G.722 (mono and stereo), and Opus.

Also adding checks on number of output channels.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 14:19:00 +00:00
kjellander@webrtc.org
7e86049d21 Roll chromium_revision 681cc8e..f0a439d (r292217:r292861)
Mainly to pick up https://codereview.chromium.org/500423004/
that enables us to build the Android APK tests from
a standalone checkout.

Other changes:
* tools/swarming_client to e7d8b988423ff1966d64db3ef7ca766296f9b0c1
* third_party/boringssl to 6c7aed048ca0a335e02dfee10976c5dc8620783e
* third_party/icu 527ea2dd86afa2751a85d1cc4695f9e2e2d18022 (r291706)
* third_party/libjpeg_turbo to 2ed5319 (r291725)
* third_party/libvpx 563c46b:982d147 (r291661:r291730)
* third_party/nss to 90c5f9a8b8980fe60165813f578bbeb4fe20b18d

Trybot failures at Android trybots are expected, since
they're currently in a bad state since they in the middle
of being reconfigured, partially pending this CL.

BUG=webrtc:3741
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7015 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 11:41:56 +00:00
kjellander@webrtc.org
3bd4156d75 Android APK tests built from a normal WebRTC checkout.
Restructure how the Android APK tests are compiled now
that we have a Chromium checkout available (since r6938).

This removes the need of several hacks that were needed when
building these targets from inside a Chromium checkout.
By creating a symlink to Chromium's base we can compile the required
targets. This also removes the need of the previously precompiled
binaries we keep in /deps/tools/android at Google code.

All the user needs to do is to add the target_os = ["android"]
entry to his .gclient as described at
https://code.google.com/p/chromium/wiki/AndroidBuildInstructions

Before committing this CL, the Android APK buildbots will need
to be updated.
This also solves http://crbug.com/402594 since the apply_svn_patch.py
usage will be similar to the other standalone bots.
It also solves http://crbug.com/399297

BUG=chromium:399297, chromium:402594
TESTED=Locally compiled all APK targets by running:
GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release

checkdeps

R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 11:06:37 +00:00
kjellander@webrtc.org
c4870bb221 GN: Audio device module
The GN files are based upon the GYP files as of r7009.

BUG=3441
TESTED=Passing builds with:
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/14259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
kjellander@webrtc.org
524b8f7304 GN: Implement voice engine, common audio, audio coding and audio processing
NOTICE: Assembly offsets generation for audio processing will
not be ported to GN and the process of removing them is tracked
in https://code.google.com/p/webrtc/issues/detail?id=3580.

The GN files are based upon the GYP files as of r7009.

BUG=3441
TESTED=Passing builds with:
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 aec_debug_dump=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=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

I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now:
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && 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=andrew@webrtc.org, brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
kjellander@webrtc.org
1b9a188ba5 GN: Fix webrtc/video/BUILD.gn for Chromium build.
A mistake was made in https://review.webrtc.org/18709004/
so it doesn't build in Chromium. Adding a config to get
the root folder included in the include path solves it.

BUG=3441
TESTED=Local compilation of Chromium's all target with
src/third_party/webrtc linked to the WebRTC checkout with
this CL applied.
TBR=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7011 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 21:39:35 +00:00
andrew@webrtc.org
a22485eaf0 MIPS optimizations for AEC audio processing module
Added new optimizations for MIPS that were removed in r6797.
For more information about this see https://code.google.com/p/webrtc/source/detail?r=6797

R=andrew@webrtc.org, djordje.pesut@imgtec.com

Review URL: https://webrtc-codereview.appspot.com/15259004

Patch from Ljubomir Papuga <ljubomir.papuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7010 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:51:28 +00:00