Commit Graph

5834 Commits

Author SHA1 Message Date
kjellander@webrtc.org
0a035c8924 Disable tests in common_video_unittests for Dr Memory.
These tests takes about 1000 seconds to execute under
Dr Memory Full, causing them to be killed if taking more
than 1200 seconds.
* TestScaler.BiLinearScaleTest
* TestScaler.BoxScaleTest

BUG=3247
TBR=zhaoqin@google.com

Review URL: https://webrtc-codereview.appspot.com/12289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5951 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:07:03 +00:00
henrik.lundin@webrtc.org
372ae83228 Reland "Make VoiceEngine choose ACM2 by default""
This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 07:21:03 +00:00
bjornv@webrtc.org
5964fe0f86 audio_processing: DestroyHandle() now returns void
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.

BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 06:52:28 +00:00
bjornv@webrtc.org
2a796720f8 common_audio: VADFree() now returns void
* Files in audio_coding are not affected since they never use the return value.
* voice_detection in audio_processing does.
* Updated vad_unittest.cc

BUG=441
TESTED=trybots
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5948 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 04:45:35 +00:00
fbarchard@google.com
3dfabf928c libyuv r1000 roll for DEPS update to new chromium moving location of gold linker on linux.
BUG=libyuv:323
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12049006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-21 22:06:06 +00:00
tkchin@webrtc.org
ec3d8ecdcc Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-21 18:47:24 +00:00
andrew@webrtc.org
229e16e254 Add resource audio for audio processing tests.
This is a prerequisite of:
http://review.webrtc.org/9919004/

TBR=bjornv
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-20 03:54:46 +00:00
mallinath@webrtc.org
54fd70046d Remove ASSERT in TransportChannelProxy::SetImplementation, when
proxy already set to same transport channel impl.

Since session can call SetImplementation multiple times with or without BUNDLE, there are cases when SetImplementation is called with same impl (OnRemoteCandidates/PushdownTransportDescription/SetupMux). Also variables in
cricket::TransportProxy like |connecting_| and |negotiated_| are accessed
both between worker thread and signaling threads (which calls for bigger change
on how session interacts with Transport and TransportChannelProxy). I have a created a separate bug to address later issue.

Also if single thread used as worker and signaling thread, we can end up
calling SetLocalDescription and OnRemoteCandidates in same call sequence, which
will end up calling SetImplementation twice.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5944 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 01:03:33 +00:00
andrew@webrtc.org
f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00
sergeyu@chromium.org
3d9ec1fed4 Fix multi-monitor support in the screen capturer for Mac.
This feature was broken in r5471.

BUG=361919
R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=5937

Review URL: https://webrtc-codereview.appspot.com/12109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:25:35 +00:00
buildbot@webrtc.org
8e5ec52e76 (Auto)update libjingle 65152644-> 65219629
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:00:31 +00:00
sergeyu@chromium.org
7d055a6e63 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac."
This would break when rolled in chromium because some code in
chromium depends on the code I changed in that change.

TBR=jiayl@webrtc.org
BUG=361919

Review URL: https://webrtc-codereview.appspot.com/12199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 23:45:38 +00:00
andrew@webrtc.org
0daa8be9d6 Add Chromium's ScopedVector.
Trivial changes from the original excepting scoped_vector_unittest.cc,
diff here: https://paste.googleplex.com/6664017300946944

This is a prerequisite for:
http://review.webrtc.org/9919004/

TBR=henrike@webrtc.org
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 21:20:54 +00:00
sergeyu@chromium.org
be7585b150 Fix multi-monitor support in the screen capturer for Mac.
This feature was broken in r5471.

BUG=361919
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 18:22:41 +00:00
turaj@webrtc.org
a596a389ea Fix iSAC/48000 issue with ACM2.
Registeration of iSAC into NetEq is through injecting and external AudioDecoder. This is because iSAC encoder and decoder need to share instances for bandwidth estimator to work. When external decoder is registerred, the sampling rate of the decoder had to be specified. iSAC/48000 decoder has a native sampling rate of 32000 Hz, but it has been registered as 48000 Hz decoder.

This CL fixing this issue by letting NetEq to obtain sampling rate of an external coder according to its existing database.

BUG=3143
TEST=voe_cmd_test,modules_unittest,try-bots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 23:30:49 +00:00
fischman@webrtc.org
29540b1879 Revert "PeerConnectionFactory: delay deletion of owned threads."
This reverts r5933 because it broke
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598

BUG=3100

Review URL: https://webrtc-codereview.appspot.com/12159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:54:30 +00:00
buildbot@webrtc.org
1a87f529a2 (Auto)update libjingle 65151416-> 65151642
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:41:30 +00:00
fischman@webrtc.org
cea024d672 PeerConnectionFactory: delay deletion of owned threads.
Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's
possible for ~PeerConnectionFactory() to be run on its signaling thread.
Deleting a thread from within that thread is sad times, so don't do it.

It would be nicer to avoid having PeerConnection hold a ref to the factory,
and instead require the user to keep the factory alive.  Unfortunately that
changes the contract on PeerConnection{,Factory} and it's unclear how to vet
existing callers for safety.

BUG=3100
R=juberti@webrtc.org, noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/11289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:36:00 +00:00
marpan@webrtc.org
b476d36120 Roll libvpx 259973:264320
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/12069007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 20:35:03 +00:00
henrike@webrtc.org
aeb0c28193 Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES".
BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 14:15:43 +00:00
kwiberg@webrtc.org
e57ae02327 WebRtcAecm_Process: Reduce code duplication
BUG=
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:28:33 +00:00
kwiberg@webrtc.org
d2f366f28c StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
The max value is ((2**15 - 1) + (2**15 - 1)) >> 1
              == (2**16 - 2) >> 1
              == 2**15 - 1
which doesn't overflow.

The min value is (-2**15 + -2**15) >> 1
              == -2**16 >> 1
              == -2**15
which doesn't overflow.

Since those two bracket all possible results, the call to
WebRtcSpl_SatW32ToW16 is redundant.

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

Review URL: https://webrtc-codereview.appspot.com/12019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:17:39 +00:00
henrika@webrtc.org
66803489f9 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:45:01 +00:00
henrika@webrtc.org
0f7375504a Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:38:08 +00:00
henrik.lundin@webrtc.org
e2e9abb3bc Revert "Make VoiceEngine choose ACM2 by default"
The reason for reverting is that Issue 3143 should be resolved
first.

TBR=henrika@webrtc.org
BUG=3143

Review URL: https://webrtc-codereview.appspot.com/12119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:12:27 +00:00
buildbot@webrtc.org
0b3c6c3191 (Auto)update libjingle 65086785-> 65104022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:03:57 +00:00
henrik.lundin@webrtc.org
adaf809612 Removing AudioCoding duplicate tests
Reverting to using one version of ACM in ACM tests.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:29:10 +00:00
henrik.lundin@webrtc.org
6cec07f6a7 Make VoiceEngine choose ACM2 by default
The use of a factory for ACM will be removed in later CLs.

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:07:18 +00:00
fischman@webrtc.org
c0a15b7ddc Fix crashes due to dangling external decoder pointer.
When checking whether we need to release external decoder,
we have to do pointer comparison. We can't rely on payload
types, because payload types can be stale (e.g. before we
decode the first video frame after RegisterReceiveCodec).
This leaves a dangling pointer to external decoder, which
leads to crashes later, after we actually delete the
external decoder object.
This change has been verified in Chromecast code tree.

BUG=chromium:335539
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12049004

Patch from Sergey Volk <servolk@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 01:22:48 +00:00
buildbot@webrtc.org
39b868bad3 (Auto)update libjingle 65055925-> 65086785
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 00:04:39 +00:00
jiayl@webrtc.org
8f88f20af2 Expand the test max wait time from 1000ms to 2000ms.
The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.

BUG=2838
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 17:14:21 +00:00
kjellander@webrtc.org
c18729165a Set include_internal_video_capture=1 for video_capture_tests
Having this override in the .gypi file avoids having to set it for the bots, which I think is best if we can avoid.

This CL also reverts r5869 so video_capture_tests are compiled for Android again.

BUG=2974,3152
TEST=Successfully ran:
git try -t compile
git try --bot=win_baremetal,mac_baremetal,linux_baremetal -t video_capture_tests
git try --bot=android_apk,android_apk_rel
Verified the change in webrtc/build/apk_tests.gyp makes the build compile successfully from a Chromium+WebRTC configured checkout for Android APK tests.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 12:59:49 +00:00
aluebs@webrtc.org
f927fd6481 Re-enable AGC tests:
* AgcConfigTest.HasCorrectDefaultConfiguration
* AgcConfigTest.DealsWithInvalidParameters
* AgcConfigTest.CanGetAndSetAgcStatus
* AgcConfigTest.HasCorrectDefaultRxConfiguration
* AgcConfigTest.DealsWithInvalidRxParameters
* AgcConfigTest.CanGetAndSetRxAgcStatus
* AudioProcessingTest.AgcIsOnByDefault
* AudioProcessingTest.CanEnableAgcWithAllModes
* AudioProcessingTest.RxAgcShouldBeOffByDefault
* AudioProcessingTest.CanTurnOnDigitalRxAcg
* AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc

BUG=webrtc:2784
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 11:58:18 +00:00
kjellander@webrtc.org
7de47bce12 Remove use of tmpnam.
This solves compilation with the Mac SDK 10.9.

BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org, henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
andrew@webrtc.org
2c3f1abb69 Replace flooding logs in rtp_sender.cc with a comment.
Started occurring after:
https://webrtc-codereview.appspot.com/11129004

BUG=3153
R=andresp@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 21:26:34 +00:00
wu@webrtc.org
36eda7cf0e Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end.
For example:
"m=application 38233 DTLS/SCTP 5000 "

BUG=3212
TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:37:30 +00:00
fischman@webrtc.org
ca539bbed0 iOS: baby steps to being able to include_tests=1
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
  hardware_before_streaming_test.cc

BUG=3185
R=kjellander@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:26:41 +00:00
henrik.lundin@webrtc.org
7c6e3d188a Moved voe_neteq_stats_unittest to audio_coding_module_unittest
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.

BUG=2996
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:59:25 +00:00
wu@webrtc.org
6c75c98964 Propagate capture ntp timestamp from rtp to renderer.
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.

TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:46:33 +00:00
buildbot@webrtc.org
1fd5b45a0e (Auto)update libjingle 64956819-> 64982143
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:39:43 +00:00
stefan@webrtc.org
2f8d5f3302 Check if a header extension is registered before updating it and fail silently if it's not.
BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 12:28:46 +00:00
kjellander@webrtc.org
190b72a350 Make libjingle Android example build without sourcing envsetup.sh
See https://webrtc-codereview.appspot.com/11799004
for full details (separate to avoid webrtc+talk changes in same CL).

BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 08:35:49 +00:00
kjellander@webrtc.org
6e105ede38 Make WebRTC Android examples build without sourcing envsetup.sh
The new recipes framework for configuring build explicitly sets the
GYP_DEFINES for Android builds instead of relying on the envsetup.sh script
which probably will be removed at some point in the future.

This causes our build to break since our Android examples relies on the
Android SDK being found using the ANDROID_SDK_ROOT environment variable.
A GYP variable 'android_sdk_root' exists and is set correctly by
common.gypi, which is what I'm using to pass this path correctly to these
tests.

The libjingle example is handled separately in
https://webrtc-codereview.appspot.com/11809004/

BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 08:35:00 +00:00
mallinath@webrtc.org
ad4440a64e In shared socket mode, use udp port as default receiver even if
stun server address is not set.

This can happen in a loopback scenarios where clients do not need
to provide any server information.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 01:10:58 +00:00
buildbot@webrtc.org
505f400f27 (Auto)update libjingle 64909599-> 64919255
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:33:47 +00:00
fischman@webrtc.org
e98598d3f0 Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of
adding or renaming files. If you're doing\n# structural changes, please get a
review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >>
$d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the non-talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5904 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:31:16 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
henrik.lundin@webrtc.org
35ead381f8 Adding a config struct to NetEq
With this change, the parameters sent to the NetEq::Create method are
collected in one NetEq::Config struct. The benefit is that it is easier
to set, change and override default values, and easier to expand with
more parameters in the future.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:49:17 +00:00
henrik.lundin@webrtc.org
810acbc93e New Packet and PacketSource classes for NetEq tests
These new classes are intended to replace the old NETEQTEST_RTPpacket
classes. The code in rtp_analyze.cc has been updated to use the new
classes; other test applications will follow.

BUG=2692
R=andrew@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:42:23 +00:00
buildbot@webrtc.org
1da6047132 (Auto)update libjingle 64813990-> 64909599
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:15:15 +00:00