2525 Commits

Author SHA1 Message Date
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
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
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
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
andrew@webrtc.org
af7fdfcde8 Add LTO support for Android Chromium.
This is to add support for a Link-Time Optimizations experiment in Android Chromium. As it is disabled by default, it won't change anything for most configurations.
BUG=chromium:407544
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:41:13 +00:00
henrik.lundin@webrtc.org
f554d75288 Allow same src and dst in InputAudioFile::DuplicateInterleaved
This change allows the input and output to the static method
InputAudioFile::DuplicateInterleaved to be the same array. That is,
in-place manipulation is now possible. A unit test is also added.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7008 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 07:26:40 +00:00
andrew@webrtc.org
b0dc3d7204 Precompile out our standalone CHECK macros in a Chromium build.
As documented, the use of overrides/webrtc/base/logging.h in a Chromium
build reuslts in redefined macro errors. Fortunately, Chromium's macros
can be used as drop-in replacements for the standalone versions.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/17239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7004 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 19:00:15 +00:00
andrew@webrtc.org
a5b7869f3d Add CHECK and friends from Chromium.
Replace FATAL_ERROR_IF with the more familiar (to Chromium developers)
CHECK and DCHECK. The full Chromium implementation is fairly elaborate
but I copied enough to get us most of the benefits. I believe the main
missing component is a more advanced stack dump. For this bit I relied
on the V8 implementation.

There are a few minor modifications from the Chromium original:
- The FatalMessage class is specialized for logging fatal error
messages and aborting. Chromium uses the general LogMessage class,
which we could consider moving towards in the future.
- NOTIMPLEMENTED() and NOTREACHED() have been removed, partly because
I don't want to rely on our logging.h until base/ and system_wrappers/
are consolidated.
- FATAL() replaces LOG(FATAL).

Minor modifications from V8's stack dump:
- If parsing of a stack trace symbol fails, just print the unparsed
symbol. (I noticed this happened on Mac.)
- Use __GLIBCXX__ and __UCLIBC__. This is from examining the backtrace
use in Chromium.

UNREACHABLE() has been removed because its behavior is different than
Chromium's NOTREACHED(), which is bound to cause confusion. The few uses
were replaced with FATAL(), matching the previous behavior.

Add a NO_RETURN macro, allowing us to remove unreachable return
statements following a CHECK/FATAL.

TESTED=the addition of dummy CHECK, DCHECK, CHECK_EQ and FATAL did the
did the right things. Stack traces work on Mac, but I don't get symbols
on Linux.

R=henrik.lundin@webrtc.org, kwiberg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7003 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 16:28:26 +00:00
jiayl@webrtc.org
11c6bde474 Specify an ECDH group for ECDHE.
By default, OpenSSL cannot negotiate ECDHE cipher suites as a server because it
doesn't know what curve to use.

BUG=chromium:406458
TEST=Download Firefox nightly build from 2014-08-12.
  https://ftp.mozilla.org/pub/mozilla.org/firefox/nightly/2014-08-12-mozilla-central-debug/
  Point Firefox to https://apprtc.appspot.com
  Point Chrome on Android to the URL Firefox redirects to (it'll say ?r=NUMBERS at the end)
  After tapping through various permissions prompts on either side, the call goes through.

R=agl@chromium.org, henrike@webrtc.org, jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7002 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 16:14:38 +00:00
henrike@webrtc.org
55e9da1772 Add talk owners to migrated talk folders
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7001 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 16:03:58 +00:00
kjellander@webrtc.org
788f0581c7 GN: Implement video_engine, video_capture and video_render.
Also add more from common.gypi to webrtc.gni.

These GN configs are based on GYP files in r6997.

BUG=3441
TEST=Trybots and local compile using:
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

Passed compile from a Chromium checkout with src/third_party/webrtc linked to the webrtc/ dir of a checkout with this patch applied.

R=brettw@chromium.org, glaznev@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6999 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 13:51:08 +00:00
bjornv@webrtc.org
df9fef6638 common_audio: Removed macro WEBRTC_SPL_DIV
The macro has no built-in divide by zero check. The only thing that is done is casting to int32_t.
In addition a bug was discovered where it was supposed to do a division with rounding, but instead did a division with truncation + addition by 2. This is corrected in this CL.

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 12:57:32 +00:00
aluebs@webrtc.org
59a1b1b928 Fix the different samples per channel in aecdump
BUG=webrtc:3359
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6996 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 10:43:09 +00:00
asapersson@webrtc.org
f8723d666a Add unit tests to rtcp_receiver_test.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6994 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 07:35:06 +00:00
henrik.lundin@webrtc.org
18584fcde4 Move end of namespace inside #ifdef
The code did not compile unless WEBRTC_ANDROID was defined.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6989 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 10:17:22 +00:00
andresp@webrtc.org
c3c29113d1 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module
to set the payload type to be used without having to call SendOutgoingData.

BUG=3694
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 09:39:43 +00:00
henrike@webrtc.org
66a3582170 Create a copy of talk/sound under webrtc/sound.
BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 22:04:04 +00:00
aluebs@webrtc.org
b623c5c251 Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky
BUG=webrtc:3745
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 14:22:51 +00:00
pbos@webrtc.org
c9b3f77e65 Fix data races in VideoAdapterTest.
Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).

When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.

The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.

Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.

R=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/22169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 12:33:18 +00:00
bjornv@webrtc.org
4f71e22bf9 Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV
This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use.
This CL replaces the macro with '/' at place.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 10:25:10 +00:00
bjornv@webrtc.org
1de0cc4079 common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() optimizations on armv7
According to the issue, common_audio_unittests failed on armv7. It currently pass, so we should turn it on again. There is no print out in the issue, so the cause of failure is unknown.

BUG=740
TESTED=locally on N7
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 09:36:25 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files.
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
kjellander@webrtc.org
b96ea2aab5 Remove former team members from OWNERS and WATCHLISTS
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 06:12:08 +00:00
kjellander@webrtc.org
c23923447c Roll chromium_revision 289723:291647
To pick up recent fixes after the Chromium Git switch.

Relevant changes pulled in by this roll:
* r291168 refactor sanitizer_options (we can now remove some hacks)
* r291647 roll of nss.gyp (its paths work with how we build for iOS).

BUG=2863,3731
R=iannucci@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6967 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:16:32 +00:00
kjellander@webrtc.org
42ee5b54b5 GN: Disable Chromium clang plugins for standalone build.
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.

The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.

BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
bjornv@webrtc.org
926707b167 Refactoring common_audio: Replace trivial multiplication macro
This multiplication macro literally use the '*' operator, so there is no need for it.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 11:42:42 +00:00
bjornv@webrtc.org
d32c4389ac Re-landing r6961
common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8

This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 11:19:05 +00:00
bjornv@webrtc.org
4a616be12b Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..."
> common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8
> 
> This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.
> 
> BUG=3348,3353
> TESTED=locally on linux and trybots
> R=kwiberg@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16359004

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 10:32:22 +00:00
bjornv@webrtc.org
4f01017e2d common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8
This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 10:23:22 +00:00
bjornv@webrtc.org
6e71d17bc9 Refactoring common_audio/signal_processing: Replaces trivial macros
The macros WEBRTC_SPL_ADD_SAT_W16 and WEBRTC_SPL_ADD_SAT_W32 make direct use of the corresponding functions WebRtcSpl_AddSatW16() and WebRtcSpl_AddSatW32().
This CL replaces these macros in the code.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 07:44:52 +00:00
kwiberg@webrtc.org
584cd8da4b Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)
And in the process, make it dump WAV files instead of raw PCM.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 06:26:04 +00:00
niklas.enbom@webrtc.org
153c6162d2 Landing issue 15189004
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6951 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 14:49:28 +00:00
henrik.lundin@webrtc.org
038cee2401 Add send-side bit-exactness test for AudioCoding Module
This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:

test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest

The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.)  Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.

The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 08:59:14 +00:00
henrik.lundin@webrtc.org
9b8102cf0e Use a deterministic input in NetEqBgnTest
This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.

The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)

BUG=3715
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 08:27:44 +00:00
bjornv@webrtc.org
6b2659c660 Refactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_RSFT32BI
The WEBRTC_SPL_MUL_32_32_RSFT32BI macro was removed in r6169, since it was unused. This CL removes the arm and mips optimizations of it.

BUG=3348, 3353
TESTED=locally and trybots
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 06:13:57 +00:00
thakis@chromium.org
905f9efbae Fix clang -Wformat warnings.
An unsigned int was passed through %lu instead of %u (harmless on 32bit).
More seriously, a wide string was passed through %s, which means only the
first byte in the string got printed (since the 2nd byte is likely 0 in
UCS-2). Use %ls to include the whole string, even though it might not be
renderable in the target 8bit buffer.

BUG=chromium:82385
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 02:23:30 +00:00
thakis@chromium.org
add54ad770 Convert nsx_core_neon.S to unified syntax.
That way, it builds with both gcc and clang's integrated assembler.
No intentional behavior change.

BUG=chromium:124610
R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/15199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 02:23:26 +00:00
tnakamura@webrtc.org
8dcf61f946 Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome.
R=niklas.enbom@webrtc.org
TBR=niklas.embom

Review URL: https://webrtc-codereview.appspot.com/15219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 22:44:18 +00:00
kjellander@webrtc.org
8925662318 Make WebRTC work with Chromium Git checkouts
WebRTC standalone shares a lot of dependencies and build
tools with Chromium. To make the build work, many of the
paths of a Chromium checkout is now emulated by creating
symlinks to files and directories.

All DEPS entries that previously used Var("chromium_trunk")
to reference a Chromium checkout or From("chromium_deps"..)
to reference the Chromium DEPS file are now removed and
replaced by symlink entries in setup_links.py.

The script also handles cleanup of the legacy
Subversion-based dependencies that's needed for the
transition.

Windows: One Windows-specific important change is that
gclient sync|runhooks must now be run from a shell
with Administrator privileges in order to be able to create
symlinks. This also means that Windows XP is no longer
supported.

To transition a previously created checkout:
Run "python setup_links.py --force" to cleanup the old
SVN-based dependencies that have been synced by gclient sync.
For Buildbots, the --force flag is automatically enabled for
their syncs.

BUG=2863, chromium:339647
TEST=Manual testing on Linux, Mac and Windows.
R=andrew@webrtc.org, iannucci@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 12:10:11 +00:00
henrik.lundin@webrtc.org
3fb2d0cd0e Add TSAN suppression for heap-use-after-free in libvpx
BUG=3671
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 11:07:29 +00:00