17454f79dc
Add ctors to ChannelBuffer to enable copying on construction.
...
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.
R=claguna@google.com
Review URL: https://webrtc-codereview.appspot.com/24469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:27:04 +00:00
c64246f42c
Set a default speech type in iSAC wrapper
...
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.
BUG=crbug/411162
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:40:58 +00:00
ed8bcd3ac5
Starting to implement the new ACM API
...
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.
This is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:13:19 +00:00
9600519147
Adding the ability to test on Chrome for Android.
...
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:01:40 +00:00
37c39f3784
audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16
...
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.
Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:21:56 +00:00
0d394f3609
video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16
...
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:19:39 +00:00
c77e4d6aef
- Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices.
...
- Select BotType using nodeJs terminal command.
- ping_pong.js test added.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 10:36:11 +00:00
fe16167507
Fix RTT calculations for send-only channels.
...
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.
BUG=3781
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:45:25 +00:00
c30e9e2300
Ignore FEC packet in stats, if it is first packet on ssrc.
...
BUG=chrome:410456
R=mflodman@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:20:18 +00:00
6d08ca6379
GN: Prefix WebRTC specific variables with "rtc_"
...
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
f68cf93e1b
Add video_capture_tests_apk_target
...
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.
BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:35:51 +00:00
a781f68712
Fix rm command for class cleanup in r7091
...
In https://webrtc-codereview.appspot.com/20339004
the rm command was missing 'r' for recursive mode.
TBR=henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/26379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06 22:11:28 +00:00
9510022e1f
Cleanup temporary class files for OpenSlDemo
...
I've seen tryjobs failing when they shouldn't on
the Android trybots and I suspect this might have
something to do with it.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06 18:03:45 +00:00
8f073c5054
Create a new interface for AudioCodingModule
...
This is a first draft of the interface, and is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:16:23 +00:00
fed47dc205
Drop buildbot_tests.py script
...
This is no longer used since the buildbots have moved
over to recipes (where these arguments are configured).
See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73
for details.
This is essentially a revert of
https://webrtc-codereview.appspot.com/1021006
BUG=None
TESTED=None
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 08:25:38 +00:00
bcf75e32a3
Modifying audio_coding/codecs/OWNERS
...
Adding myself.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 07:18:50 +00:00
c2c4117477
common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing
...
Affected components:
* AECMobile
- Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)
BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 06:01:53 +00:00
f1427c6731
Revert 7070 "TurnPort should retry allocation with a new address on error
...
STUN_ERROR_ALLOCATION_MISMATCH."
TBR=jiayl@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/15359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 22:21:33 +00:00
574f2f60fe
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
...
BUG=3570
R=juberti@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 19:11:34 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
48f2568d89
audio_processing/nsx: Bug fix that could cause divide by zero
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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
34a6764981
Remove the checks.h dependence on logging.h in a standalone build.
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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
8e24d87778
Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking.
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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
d72a7599d4
Create a copy of talk/xmllite under webrtc/xmllite.
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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
6f729e8a74
Disable video_engine_tests and webrtc_perf_tests on Android.
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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
ee0fb187a5
Divide-by-zero problem in NetEq's Normal::Process fixed
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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
