external/webrtc.git
30 hours agoAdd resource audio for audio processing tests. master
andrew@webrtc.org [Sun, 20 Apr 2014 03:54:46 +0000 (03:54 +0000)]
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

2 days agoRemove ASSERT in TransportChannelProxy::SetImplementation, when
mallinath@webrtc.org [Sat, 19 Apr 2014 01:03:33 +0000 (01:03 +0000)]
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

2 days agoResampler modifications in preparation for arbitrary audioproc rates.
andrew@webrtc.org [Sat, 19 Apr 2014 00:32:07 +0000 (00:32 +0000)]
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

2 days agoFix multi-monitor support in the screen capturer for Mac.
sergeyu@chromium.org [Sat, 19 Apr 2014 00:25:35 +0000 (00:25 +0000)]
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

2 days ago(Auto)update libjingle 65152644-> 65219629
buildbot@webrtc.org [Sat, 19 Apr 2014 00:00:31 +0000 (00:00 +0000)]
(Auto)update libjingle 65152644-> 65219629

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRevert r5937 "Fix multi-monitor support in the screen capturer for Mac."
sergeyu@chromium.org [Fri, 18 Apr 2014 23:45:38 +0000 (23:45 +0000)]
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

2 days agoAdd Chromium's ScopedVector.
andrew@webrtc.org [Fri, 18 Apr 2014 21:20:54 +0000 (21:20 +0000)]
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

2 days agoFix multi-monitor support in the screen capturer for Mac.
sergeyu@chromium.org [Fri, 18 Apr 2014 18:22:41 +0000 (18:22 +0000)]
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

3 days agoFix iSAC/48000 issue with ACM2.
turaj@webrtc.org [Thu, 17 Apr 2014 23:30:49 +0000 (23:30 +0000)]
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

3 days agoRevert "PeerConnectionFactory: delay deletion of owned threads."
fischman@webrtc.org [Thu, 17 Apr 2014 22:54:30 +0000 (22:54 +0000)]
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

3 days ago(Auto)update libjingle 65151416-> 65151642
buildbot@webrtc.org [Thu, 17 Apr 2014 22:41:30 +0000 (22:41 +0000)]
(Auto)update libjingle 65151416-> 65151642

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoPeerConnectionFactory: delay deletion of owned threads.
fischman@webrtc.org [Thu, 17 Apr 2014 22:36:00 +0000 (22:36 +0000)]
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

3 days agoRoll libvpx 259973:264320
marpan@webrtc.org [Thu, 17 Apr 2014 20:35:03 +0000 (20:35 +0000)]
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

3 days agoUpdate PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES".
henrike@webrtc.org [Thu, 17 Apr 2014 14:15:43 +0000 (14:15 +0000)]
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

3 days agoWebRtcAecm_Process: Reduce code duplication
kwiberg@webrtc.org [Thu, 17 Apr 2014 12:28:33 +0000 (12:28 +0000)]
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

3 days agoStereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
kwiberg@webrtc.org [Thu, 17 Apr 2014 12:17:39 +0000 (12:17 +0000)]
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

3 days agoRemoves parts of the VoEBase sub API as part of a clean-up operation where the goal...
henrika@webrtc.org [Thu, 17 Apr 2014 10:45:01 +0000 (10:45 +0000)]
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

3 days agoRemoves VoECodec sub API as part of a clean-up operation where the goal is to remove...
henrika@webrtc.org [Thu, 17 Apr 2014 10:38:08 +0000 (10:38 +0000)]
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

4 days agoRevert "Make VoiceEngine choose ACM2 by default"
henrik.lundin@webrtc.org [Thu, 17 Apr 2014 10:12:27 +0000 (10:12 +0000)]
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

4 days ago(Auto)update libjingle 65086785-> 65104022
buildbot@webrtc.org [Thu, 17 Apr 2014 10:03:57 +0000 (10:03 +0000)]
(Auto)update libjingle 65086785-> 65104022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoRemoving AudioCoding duplicate tests
henrik.lundin@webrtc.org [Thu, 17 Apr 2014 08:29:10 +0000 (08:29 +0000)]
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

4 days agoMake VoiceEngine choose ACM2 by default
henrik.lundin@webrtc.org [Thu, 17 Apr 2014 08:07:18 +0000 (08:07 +0000)]
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

4 days agoFix crashes due to dangling external decoder pointer.
fischman@webrtc.org [Thu, 17 Apr 2014 01:22:48 +0000 (01:22 +0000)]
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

4 days ago(Auto)update libjingle 65055925-> 65086785
buildbot@webrtc.org [Thu, 17 Apr 2014 00:04:39 +0000 (00:04 +0000)]
(Auto)update libjingle 65055925-> 65086785

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoExpand the test max wait time from 1000ms to 2000ms.
jiayl@webrtc.org [Wed, 16 Apr 2014 17:14:21 +0000 (17:14 +0000)]
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

4 days agoSet include_internal_video_capture=1 for video_capture_tests
kjellander@webrtc.org [Wed, 16 Apr 2014 12:59:49 +0000 (12:59 +0000)]
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

4 days agoRe-enable AGC tests:
aluebs@webrtc.org [Wed, 16 Apr 2014 11:58:18 +0000 (11:58 +0000)]
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

5 days agoRemove use of tmpnam.
kjellander@webrtc.org [Wed, 16 Apr 2014 08:04:26 +0000 (08:04 +0000)]
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

5 days agoReplace flooding logs in rtp_sender.cc with a comment.
andrew@webrtc.org [Tue, 15 Apr 2014 21:26:34 +0000 (21:26 +0000)]
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

5 days agoWorkaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line...
wu@webrtc.org [Tue, 15 Apr 2014 20:37:30 +0000 (20:37 +0000)]
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

5 days agoiOS: baby steps to being able to include_tests=1
fischman@webrtc.org [Tue, 15 Apr 2014 20:26:41 +0000 (20:26 +0000)]
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

5 days agoMoved voe_neteq_stats_unittest to audio_coding_module_unittest
henrik.lundin@webrtc.org [Tue, 15 Apr 2014 17:59:25 +0000 (17:59 +0000)]
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

5 days agoPropagate capture ntp timestamp from rtp to renderer.
wu@webrtc.org [Tue, 15 Apr 2014 17:46:33 +0000 (17:46 +0000)]
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

5 days ago(Auto)update libjingle 64956819-> 64982143
buildbot@webrtc.org [Tue, 15 Apr 2014 17:39:43 +0000 (17:39 +0000)]
(Auto)update libjingle 64956819-> 64982143

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoCheck if a header extension is registered before updating it and fail silently if...
stefan@webrtc.org [Tue, 15 Apr 2014 12:28:46 +0000 (12:28 +0000)]
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

6 days agoMake libjingle Android example build without sourcing envsetup.sh
kjellander@webrtc.org [Tue, 15 Apr 2014 08:35:49 +0000 (08:35 +0000)]
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

6 days agoMake WebRTC Android examples build without sourcing envsetup.sh
kjellander@webrtc.org [Tue, 15 Apr 2014 08:35:00 +0000 (08:35 +0000)]
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

6 days agoIn shared socket mode, use udp port as default receiver even if
mallinath@webrtc.org [Tue, 15 Apr 2014 01:10:58 +0000 (01:10 +0000)]
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

6 days ago(Auto)update libjingle 64909599-> 64919255
buildbot@webrtc.org [Mon, 14 Apr 2014 20:33:47 +0000 (20:33 +0000)]
(Auto)update libjingle 64909599-> 64919255

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoMake everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition.
fischman@webrtc.org [Mon, 14 Apr 2014 20:31:16 +0000 (20:31 +0000)]
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

6 days agoMake everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
fischman@webrtc.org [Mon, 14 Apr 2014 20:08:03 +0000 (20:08 +0000)]
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

6 days agoAdding a config struct to NetEq
henrik.lundin@webrtc.org [Mon, 14 Apr 2014 18:49:17 +0000 (18:49 +0000)]
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

6 days agoNew Packet and PacketSource classes for NetEq tests
henrik.lundin@webrtc.org [Mon, 14 Apr 2014 18:42:23 +0000 (18:42 +0000)]
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

6 days ago(Auto)update libjingle 64813990-> 64909599
buildbot@webrtc.org [Mon, 14 Apr 2014 18:15:15 +0000 (18:15 +0000)]
(Auto)update libjingle 64813990-> 64909599

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoiosdeviceinfo.cc: remove unnecessary file
fischman@webrtc.org [Mon, 14 Apr 2014 18:12:32 +0000 (18:12 +0000)]
iosdeviceinfo.cc: remove unnecessary file

The do-nothing implementation in this file is already present in
mobiledevicemanager.cc (shared with Android) so this isn't adding value, and
causes duplicate-symbol errors under some compilers.

BUG=3201
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5899 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix gyp for video_capture/ensure_initialized.cc.
primiano@chromium.org [Mon, 14 Apr 2014 17:26:31 +0000 (17:26 +0000)]
Fix gyp for video_capture/ensure_initialized.cc.

This is a follow-up to
https://webrtc-codereview.appspot.com/11359004
which introduced an invalid dependency in the
chromium build when building without linker GC.

BUG=2974,3152,chromium:354405
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5898 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days ago(Auto)update libjingle 64709629-> 64813990
buildbot@webrtc.org [Mon, 14 Apr 2014 16:06:21 +0000 (16:06 +0000)]
(Auto)update libjingle 64709629-> 64813990

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemoves VoECallReport sub API as part of a clean-up operation where the goal is to...
henrika@webrtc.org [Mon, 14 Apr 2014 14:12:50 +0000 (14:12 +0000)]
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoAdded a new OnMoreData() interface which will not feed the playout data to APM.
xians@webrtc.org [Mon, 14 Apr 2014 10:50:37 +0000 (10:50 +0000)]
Added a new OnMoreData() interface which will not feed the playout data to APM.

BUG=3147
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdd win_drmemory_light trybot to default trybot list.
kjellander@webrtc.org [Mon, 14 Apr 2014 08:38:27 +0000 (08:38 +0000)]
Add win_drmemory_light trybot to default trybot list.

BUG=chromium:360054
TEST=None
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5894 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoDrMemory: Excluding failing tests for Dr Memory Full
kjellander@webrtc.org [Sun, 13 Apr 2014 11:45:43 +0000 (11:45 +0000)]
DrMemory: Excluding failing tests for Dr Memory Full

PortAllocatorTest.TestEnableSharedSocketWithNat
fails in libjingle_p2p_unittest.
Example:
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/2/steps/memory%20test%3A%20libjingle_p2p_unittest/logs/stdio
Previous experience tells me that disabling only this test
case will make another one in the same test suite fail,
so I'm blanked disabling the whole test.

TBR=phoglund
BUG=3158

Review URL: https://webrtc-codereview.appspot.com/11909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5893 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoDrMemory: Excluding failing tests for Dr Memory Full
kjellander@webrtc.org [Sun, 13 Apr 2014 08:31:42 +0000 (08:31 +0000)]
DrMemory: Excluding failing tests for Dr Memory Full
BUG=3158
TEST=None
TBR=kjellander,phoglund

Review URL: https://webrtc-codereview.appspot.com/11899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5892 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoDrMemory: Excluding failing tests for Dr Memory Full
kjellander@webrtc.org [Sat, 12 Apr 2014 19:53:06 +0000 (19:53 +0000)]
DrMemory: Excluding failing tests for Dr Memory Full

The PortTest.TestLocalToTurn of libjingle_p2p_unittest
and DtmfSenderTest.InsertDtmf of libjingle_peerconnection_unittest
failed in the first run on
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/
However, I cannot reproduce on my machine, so I'm disabling all
test cases of those tests, assuming the others might fail as well.

BUG=3158
TEST=None
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/11889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5891 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFix the captured screen rect conversion.
jiayl@webrtc.org [Fri, 11 Apr 2014 22:31:15 +0000 (22:31 +0000)]
Fix the captured screen rect conversion.
device_mode.dmPosition is already relative to the primary display's top-left, while the expected value of GetScreenRect() is also relative to the primary display's top-left.

TESTED=verified on Windows single monitor capturing and cursor capturing is fixed.

BUG=https://code.google.com/p/chromium/issues/detail?id=362631
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11789006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5890 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoNetEq changes.
turaj@webrtc.org [Fri, 11 Apr 2014 18:47:55 +0000 (18:47 +0000)]
NetEq changes.

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

Review URL: https://webrtc-codereview.appspot.com/9859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoDrMemory: Suppress and exclude more tests to green up the full build.
kjellander@webrtc.org [Fri, 11 Apr 2014 14:16:27 +0000 (14:16 +0000)]
DrMemory: Suppress and exclude more tests to green up the full build.

Exclude failing tests in libjingle tests and suppress
failures in modules_unittests.

Also exclude the following extremely slow tests to make it
possible to get a reasonable execution time for the bot:
* TestScaler.PointScaleTest (81573 ms)
* TestScaler.BiLinearScaleTest (1111554 ms)
* TestScaler.BoxScaleTest (1129625 ms)
* TestVp8Impl.BaseUnitTest (762598 ms)
* VideoProcessorIntegrationTest.ProcessNoLossChangeBitRate (342149 ms)

TBR=phoglund@webrtc.org
BUG=3158, 3183, 3184
TEST=Successful runs on local Windows box:
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_p2p_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_peerconnection_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test modules_unittests --tool drmemory_full

Review URL: https://webrtc-codereview.appspot.com/11819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5888 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoCleaned up logging in video_coding.
stefan@webrtc.org [Fri, 11 Apr 2014 14:08:35 +0000 (14:08 +0000)]
Cleaned up logging in video_coding.

Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.

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

Review URL: https://webrtc-codereview.appspot.com/11169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoConvert WEBRTC_TRACE to LOG in utility.
asapersson@webrtc.org [Fri, 11 Apr 2014 07:59:43 +0000 (07:59 +0000)]
Convert WEBRTC_TRACE to LOG in utility.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days ago(Auto)update libjingle 64630087-> 64709629
wu@webrtc.org [Thu, 10 Apr 2014 16:59:16 +0000 (16:59 +0000)]
(Auto)update libjingle 64630087-> 64709629

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5884 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemove erronuous commit message from auto sync.
henrike@webrtc.org [Thu, 10 Apr 2014 14:39:38 +0000 (14:39 +0000)]
Remove erronuous commit message from auto sync.

BUG=N/A
TBR=kjellander@webrtc.org

http://webrtc-codereview.appspot.com/11639004/

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5883 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoDisable UsesTraceCallback
pbos@webrtc.org [Thu, 10 Apr 2014 14:39:22 +0000 (14:39 +0000)]
Disable UsesTraceCallback

Ongoing removal of trace code is causing UsesTraceCallback to fail,
disabling it for now.

BUG=3157
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5882 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoFix loopback test for case where no constraint is given.
andresp@webrtc.org [Thu, 10 Apr 2014 14:21:45 +0000 (14:21 +0000)]
Fix loopback test for case where no constraint is given.
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemove usage of webrtc trace in video processing modules.
asapersson@webrtc.org [Thu, 10 Apr 2014 11:30:49 +0000 (11:30 +0000)]
Remove usage of  webrtc trace in video processing modules.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoAdd ability to control peer connection constraints for the loopback test.
andresp@webrtc.org [Thu, 10 Apr 2014 09:40:16 +0000 (09:40 +0000)]
Add ability to control peer connection constraints for the loopback test.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days ago(Auto)update libjingle 64594651-> 64630087
buildbot@webrtc.org [Thu, 10 Apr 2014 06:34:32 +0000 (06:34 +0000)]
(Auto)update libjingle 64594651-> 64630087

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRemove self-assignment hacks that were added to avoid unused variable warnings.
fischman@webrtc.org [Wed, 9 Apr 2014 21:19:55 +0000 (21:19 +0000)]
Remove self-assignment hacks that were added to avoid unused variable warnings.
Instead, appear to use the variables.

BUG=3152
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5877 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMove a chatty creation log in neteq to LS_VERBOSE.
andrew@webrtc.org [Wed, 9 Apr 2014 17:48:48 +0000 (17:48 +0000)]
Move a chatty creation log in neteq to LS_VERBOSE.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5876 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRemove erronuous commit message.
henrike@webrtc.org [Wed, 9 Apr 2014 14:43:43 +0000 (14:43 +0000)]
Remove erronuous commit message.

BUG=N/A
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5875 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMake Android-APK compile in release again.
solenberg@webrtc.org [Wed, 9 Apr 2014 14:21:37 +0000 (14:21 +0000)]
Make Android-APK compile in release again.

BUG=3152
R=kjellander@webrtc.org
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5874 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoPartial revert of "Removing samples directory following move to Github"
kjellander@webrtc.org [Wed, 9 Apr 2014 13:52:24 +0000 (13:52 +0000)]
Partial revert of "Removing samples directory following move to Github"

Reason: Unfortunately we depend on AppRTC being in this location
for the bots in our Chromium WebRTC waterfalls so I'm reverting
this until we've solved that dependency.

This reverts apprtc and adapter.js from being removed in r5871.

R=phoglund@webrtc.org
TBR=dutton@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/11529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5873 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days ago(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
henrika@webrtc.org [Wed, 9 Apr 2014 13:04:12 +0000 (13:04 +0000)]
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds

Landing https://webrtc-codereview.appspot.com/11419004/ manually.

TBR=niklase
BUG=none

Review URL: https://webrtc-codereview.appspot.com/11439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRemoving samples directory following move to Github
dutton@google.com [Wed, 9 Apr 2014 09:55:54 +0000 (09:55 +0000)]
Removing samples directory following move to Github

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5871 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days ago(Auto)update libjingle 64585415-> 64594651
buildbot@webrtc.org [Wed, 9 Apr 2014 06:06:38 +0000 (06:06 +0000)]
(Auto)update libjingle 64585415-> 64594651

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoUnbreak android APK buildbots by emptying the video_capture_tests_apk target.
fischman@webrtc.org [Wed, 9 Apr 2014 02:34:50 +0000 (02:34 +0000)]
Unbreak android APK buildbots by emptying the video_capture_tests_apk target.

Needed until the bots start to specify include_internal_video_capture=1.

TBR=henrike@webrtc.org
BUG=3152

Review URL: https://webrtc-codereview.appspot.com/11479006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5869 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoVideoCaptureAndroid: support multiple frame-rates per resolution.
fischman@webrtc.org [Wed, 9 Apr 2014 01:18:32 +0000 (01:18 +0000)]
VideoCaptureAndroid: support multiple frame-rates per resolution.

Also enables running video_capture_tests_apk on the WebRTC/Chromium APK bots,
assuming GYP_DEFINES includes include_tests=1 and
include_internal_video_capture=1.
This required running VideoCaptureAndroid's camera capture on a dedicated thread, matching other platform's video_capture impls.

BUG=2974,3152
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5868 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoFix DesktopSize::is_empty() for the case when only width or only height is 0.
sergeyu@chromium.org [Wed, 9 Apr 2014 01:04:22 +0000 (01:04 +0000)]
Fix DesktopSize::is_empty() for the case when only width or only height is 0.

BUG=crbug.com/358909
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5867 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoMove output_mixer_unittest.cc to utility_unittest.cc.
andrew@webrtc.org [Tue, 8 Apr 2014 23:09:28 +0000 (23:09 +0000)]
Move output_mixer_unittest.cc to utility_unittest.cc.

This reflects a move of the tested code in:
https://webrtc-codereview.appspot.com/11019005/

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/11449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5866 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoVideoCaptureAndroid: stop referencing ViERenderer
fischman@webrtc.org [Tue, 8 Apr 2014 22:55:07 +0000 (22:55 +0000)]
VideoCaptureAndroid: stop referencing ViERenderer

To facilitate building video_capture's java code without video_render's java
code this reorganizes the local-preview hack to be driven by MediaEngine.
This is the "first step" in the linked bug.

BUG=3175
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5865 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days ago(Auto)update libjingle 64326665-> 64585415
henrike@webrtc.org [Tue, 8 Apr 2014 22:13:01 +0000 (22:13 +0000)]
(Auto)update libjingle 64326665-> 64585415

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5864 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agovideo_capture(iOS): move stopCapture to background thread
fischman@webrtc.org [Tue, 8 Apr 2014 21:06:52 +0000 (21:06 +0000)]
video_capture(iOS): move stopCapture to background thread

Also suspend frame delivery on stopCapture() to avoid pause+onVideoError
during hangup.

BUG=3162
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/11389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5863 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoImplement FEC support in VideoReceiveStream.
pbos@webrtc.org [Tue, 8 Apr 2014 11:21:45 +0000 (11:21 +0000)]
Implement FEC support in VideoReceiveStream.

Added an FEC end-to-end test. NACK+FEC is probably working but not yet tested
as the test for it must introduce packet delays as the underlying API prefers
NACK over FEC if RTT is low.

BUG=3174
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5862 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoConvert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
andresp@webrtc.org [Tue, 8 Apr 2014 11:06:12 +0000 (11:06 +0000)]
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
Clean some logs and add asserts in the way.

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

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoNew NetEq test to verify correct timestamp propagation
henrik.lundin@webrtc.org [Mon, 7 Apr 2014 21:21:45 +0000 (21:21 +0000)]
New NetEq test to verify correct timestamp propagation

BUG=3154
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5860 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoRemoves unused thread causing compiler warnings.
henrike@webrtc.org [Mon, 7 Apr 2014 20:49:34 +0000 (20:49 +0000)]
Removes unused thread causing compiler warnings.

BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5859 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoCompare the answer's media type against offer to make sure they are match. Otherwise...
wu@webrtc.org [Mon, 7 Apr 2014 17:04:35 +0000 (17:04 +0000)]
Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure.

BUG=2687
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoRemoved the disabling of include_tests from r2729.
henrike@webrtc.org [Mon, 7 Apr 2014 15:52:31 +0000 (15:52 +0000)]
Removed the disabling of include_tests from r2729.

BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5856 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoUpdated WebRTC version to 3.52
elham@webrtc.org [Mon, 7 Apr 2014 15:49:00 +0000 (15:49 +0000)]
Updated WebRTC version to 3.52
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5855 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoClean up traces and logs in RemoteBitrateEstimator.
stefan@webrtc.org [Mon, 7 Apr 2014 12:53:28 +0000 (12:53 +0000)]
Clean up traces and logs in RemoteBitrateEstimator.

BUG=3153
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5854 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoLog Fixit for parts of video_engine folder.
mflodman@webrtc.org [Mon, 7 Apr 2014 10:56:31 +0000 (10:56 +0000)]
Log Fixit for parts of video_engine folder.

BUG=3153
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5853 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoDisable more tests for DrMemory to speed up execution.
kjellander@webrtc.org [Mon, 7 Apr 2014 09:00:12 +0000 (09:00 +0000)]
Disable more tests for DrMemory to speed up execution.

Disable a few more tests on Windows when running under
Dr Memory to get the build time down to a reasonable total.

BUG=None
TEST=None
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/11299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5852 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoFix logging calls in bitrate_controller module.
andresp@webrtc.org [Mon, 7 Apr 2014 08:45:16 +0000 (08:45 +0000)]
Fix logging calls in bitrate_controller module.

BUG=3153
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5851 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoExcluding and suppressing Dr Memory test failures.
kjellander@webrtc.org [Mon, 7 Apr 2014 08:01:06 +0000 (08:01 +0000)]
Excluding and suppressing Dr Memory test failures.

With these tests excluded and failures suppressed
we should be able to bring Dr Memory Full into a
green state in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
so we can move the bots into the main waterfall.

BUG=3158, 3159
TEST=Ran successful runs of the tests that never completed
using the reproduction steps in the issues listed above on
a local Windows box. The tests that just failed weren't tried,
since they cannot have been blocking other possibly failing
tests in the same binary.

R=pbos@webrtc.org
TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/11209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5850 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoRemove WEBRTC_TRACE use in common_video/
pbos@webrtc.org [Mon, 7 Apr 2014 07:29:18 +0000 (07:29 +0000)]
Remove WEBRTC_TRACE use in common_video/

Replaces a NOTREACHED() macro with inline assert(false).

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5849 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoTalk: fixes warning: local variable is initialized but not referenced due to only...
henrike@webrtc.org [Fri, 4 Apr 2014 22:33:34 +0000 (22:33 +0000)]
Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts.

BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5848 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoAppRTCDemo(android): fix a couple of SDP-related regressions.
fischman@webrtc.org [Fri, 4 Apr 2014 21:40:46 +0000 (21:40 +0000)]
AppRTCDemo(android): fix a couple of SDP-related regressions.

- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
  opportunities for improvement in the preferISAC; changed split/join to use
  \r\n instead of \n and now omitting the trailing space on the m=audio line
  that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
  calling.  apprtc.py suppresses DTLS for that reason in loopback calls, so the
  android demo app now only enables DTLS by default if it is not suppressed by a
  constraint (matching Chrome).

BUG=3164,3165,2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoFix a crash in WindowCapturereMac when capture() fails.
jiayl@webrtc.org [Fri, 4 Apr 2014 20:26:41 +0000 (20:26 +0000)]
Fix a crash in WindowCapturereMac when capture() fails.

BUG=http://code.google.com/p/chromium/issues/detail?id=359985
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5846 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks ago(Auto)update libjingle 64247466-> 64326665
henrike@webrtc.org [Fri, 4 Apr 2014 18:39:07 +0000 (18:39 +0000)]
(Auto)update libjingle 64247466-> 64326665

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5845 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoFix the library path for android 64-bit build
michaelbai@google.com [Fri, 4 Apr 2014 04:44:19 +0000 (04:44 +0000)]
Fix the library path for android 64-bit build

BUG=359687
R=andrew@webrtc.org, fischman@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5844 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoConsolidate audio conversion from Channel and TransmitMixer.
andrew@webrtc.org [Thu, 3 Apr 2014 21:56:01 +0000 (21:56 +0000)]
Consolidate audio conversion from Channel and TransmitMixer.

Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.

Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.

BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.

R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoRemoved rehydrate.html
dutton@google.com [Thu, 3 Apr 2014 21:25:54 +0000 (21:25 +0000)]
Removed rehydrate.html

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5842 4adac7df-926f-26a2-2b94-8c16560cd09d