external/webrtc.git
5 hours agoThe lastest commit on this file was in master
minyue@webrtc.org [Tue, 22 Jul 2014 09:55:51 +0000 (09:55 +0000)]
The lastest commit on this file was in

https://webrtc-codereview.appspot.com/15529004/

The final patch set should have included this, but was missed.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d

6 hours agoEnable ReceiveStreamReceivingByDefault test.
pbos@webrtc.org [Tue, 22 Jul 2014 09:14:58 +0000 (09:14 +0000)]
Enable ReceiveStreamReceivingByDefault test.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6754 4adac7df-926f-26a2-2b94-8c16560cd09d

8 hours agoRemove no longer used SkipEncodingUnusedStreams.
andresp@webrtc.org [Tue, 22 Jul 2014 07:17:17 +0000 (07:17 +0000)]
Remove no longer used SkipEncodingUnusedStreams.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6753 4adac7df-926f-26a2-2b94-8c16560cd09d

8 hours agoRemove remains of WEBRTC_NO_STL.
andresp@webrtc.org [Tue, 22 Jul 2014 06:48:58 +0000 (06:48 +0000)]
Remove remains of WEBRTC_NO_STL.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d

17 hours ago(Auto)update libjingle 71599033-> 71605904
buildbot@webrtc.org [Mon, 21 Jul 2014 21:55:43 +0000 (21:55 +0000)]
(Auto)update libjingle 71599033-> 71605904

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6751 4adac7df-926f-26a2-2b94-8c16560cd09d

18 hours ago(Auto)update libjingle 71575585-> 71599033
buildbot@webrtc.org [Mon, 21 Jul 2014 20:38:58 +0000 (20:38 +0000)]
(Auto)update libjingle 71575585-> 71599033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d

22 hours agoMIPS optimizations for ISAC (patch #2)
andrew@webrtc.org [Mon, 21 Jul 2014 16:43:13 +0000 (16:43 +0000)]
MIPS optimizations for ISAC (patch #2)

Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32

Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19749004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d

27 hours agoDisable GetStatsForInvalidTrack while I rewrite it.
tommi@webrtc.org [Mon, 21 Jul 2014 11:44:39 +0000 (11:44 +0000)]
Disable GetStatsForInvalidTrack while I rewrite it.

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d

28 hours agoRefactor StatsCollector and associated types.
tommi@webrtc.org [Mon, 21 Jul 2014 11:24:17 +0000 (11:24 +0000)]
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6745

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d

28 hours agoRevert 6745 "Refactor StatsCollector and associated types."
tommi@webrtc.org [Mon, 21 Jul 2014 11:05:28 +0000 (11:05 +0000)]
Revert 6745 "Refactor StatsCollector and associated types."
Broke build on android.

> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18819004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d

28 hours agoRefactor StatsCollector and associated types.
tommi@webrtc.org [Mon, 21 Jul 2014 10:55:11 +0000 (10:55 +0000)]
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d

47 hours agoCheck before send/receive rtp header extensions.
pbos@webrtc.org [Sun, 20 Jul 2014 15:27:35 +0000 (15:27 +0000)]
Check before send/receive rtp header extensions.

BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13949004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoImplement Base::ConstrainNewCodec2.
pbos@webrtc.org [Sun, 20 Jul 2014 14:40:23 +0000 (14:40 +0000)]
Implement Base::ConstrainNewCodec2.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6743 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoIgnore empty data in DataChannel::Send to match FF's behavior.
jiayl@webrtc.org [Fri, 18 Jul 2014 23:57:50 +0000 (23:57 +0000)]
Ignore empty data in DataChannel::Send to match FF's behavior.

BUG=crbug/395205
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 71460499-> 71464449
buildbot@webrtc.org [Fri, 18 Jul 2014 23:31:30 +0000 (23:31 +0000)]
(Auto)update libjingle 71460499-> 71464449

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6741 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRevert "Reland r6707 with the fix for callclient.cc."
jiayl@webrtc.org [Fri, 18 Jul 2014 22:28:36 +0000 (22:28 +0000)]
Revert "Reland r6707 with the fix for callclient.cc."

Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.

TBR=wu@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/17979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 71456344-> 71456420
buildbot@webrtc.org [Fri, 18 Jul 2014 21:41:41 +0000 (21:41 +0000)]
(Auto)update libjingle 71456344-> 71456420

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6739 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 71456173-> 71456344
buildbot@webrtc.org [Fri, 18 Jul 2014 21:39:56 +0000 (21:39 +0000)]
(Auto)update libjingle 71456173-> 71456344

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoReland r6707 with the fix for callclient.cc.
jiayl@webrtc.org [Fri, 18 Jul 2014 21:34:11 +0000 (21:34 +0000)]
Reland r6707 with the fix for callclient.cc.

TBR=mallinath@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/13039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoThis is to re-open an earlier CL
minyue@webrtc.org [Fri, 18 Jul 2014 21:11:27 +0000 (21:11 +0000)]
This is to re-open an earlier CL

https://webrtc-codereview.appspot.com/16619005/

which is reverted due to an issue in audio conference mixer.

This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/

BUG=webrtc:3155
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 71452608-> 71453580
buildbot@webrtc.org [Fri, 18 Jul 2014 21:07:50 +0000 (21:07 +0000)]
(Auto)update libjingle 71452608-> 71453580

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoCreates the default track if the remote media content is send-only and there is no...
jiayl@webrtc.org [Fri, 18 Jul 2014 20:54:27 +0000 (20:54 +0000)]
Creates the default track if the remote media content is send-only and there is no stream in the SDP.

BUG=2628
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRuntime guard for iOS7 property.
tkchin@webrtc.org [Fri, 18 Jul 2014 17:17:59 +0000 (17:17 +0000)]
Runtime guard for iOS7 property.

BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6733 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoFix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
tkchin@webrtc.org [Fri, 18 Jul 2014 17:13:28 +0000 (17:13 +0000)]
Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.

BUG=3581
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoDisable GetStats on DrMemory.
pbos@webrtc.org [Fri, 18 Jul 2014 13:33:48 +0000 (13:33 +0000)]
Disable GetStats on DrMemory.

Flakes/fails on DrMemory Full just like the implementation in
webrtcvideoengine.cc.

BUG=3482
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6731 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoThis is related to an earlier CL of enabling Opus 48 kHz.
minyue@webrtc.org [Fri, 18 Jul 2014 12:28:28 +0000 (12:28 +0000)]
This is related to an earlier CL of enabling Opus 48 kHz.
https://webrtc-codereview.appspot.com/16619005/

It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.

TEST=locally solved https://webrtc-codereview.appspot.com/16619005/

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoInitial WebRtcVideoEngine2::GetStats().
pbos@webrtc.org [Fri, 18 Jul 2014 11:11:55 +0000 (11:11 +0000)]
Initial WebRtcVideoEngine2::GetStats().

Also forward-declaring and moving WebRtcVideoRenderer out of header.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoSleep in ThreadTest thread functions.
pbos@webrtc.org [Fri, 18 Jul 2014 10:12:50 +0000 (10:12 +0000)]
Sleep in ThreadTest thread functions.

Prevents busy loops that really mess up Valgrind's thread scheduling,
this brings runtimes from up to minutes down to milliseconds.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6728 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoRestart VideoReceiveStreams in WebRtcVideoEngine2.
pbos@webrtc.org [Fri, 18 Jul 2014 09:35:58 +0000 (09:35 +0000)]
Restart VideoReceiveStreams in WebRtcVideoEngine2.

Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.

Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days ago(Auto)update libjingle 71378257-> 71410012
buildbot@webrtc.org [Fri, 18 Jul 2014 08:22:39 +0000 (08:22 +0000)]
(Auto)update libjingle 71378257-> 71410012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
kwiberg@webrtc.org [Fri, 18 Jul 2014 07:50:29 +0000 (07:50 +0000)]
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float

Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoReduce runtime of RingBufferTest by a factor of 100.
andrew@webrtc.org [Thu, 17 Jul 2014 23:16:44 +0000 (23:16 +0000)]
Reduce runtime of RingBufferTest by a factor of 100.

This test was needlessly long.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/15029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6724 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoUse _numMixedParticipants instead of audioFrameList->size() to determine if there...
wu@webrtc.org [Thu, 17 Jul 2014 22:19:21 +0000 (22:19 +0000)]
Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants.

There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.

TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoConnect to the turn server if address cannot be resolved by the browser by using
mallinath@webrtc.org [Thu, 17 Jul 2014 21:55:04 +0000 (21:55 +0000)]
Connect to the turn server if address cannot be resolved by the browser by using
unresolved address. This case is only considered for TCP sockets. P2P layer will
assume socket will do the resolve by using a proxy.

BUG=3384
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6722 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAssigning a priority to TURN server list passed to PeerConnection. First entry in...
mallinath@webrtc.org [Thu, 17 Jul 2014 18:23:52 +0000 (18:23 +0000)]
Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.

This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.

BUG=3223
R=jiayl@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agofix
jiayl@webrtc.org [Thu, 17 Jul 2014 17:07:49 +0000 (17:07 +0000)]
fix

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix issue where padding is sent before media with undefined timestamps if not abs...
stefan@webrtc.org [Thu, 17 Jul 2014 16:10:14 +0000 (16:10 +0000)]
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.

This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove unused ExperimentalNS API in AudioProcessing
aluebs@webrtc.org [Thu, 17 Jul 2014 11:32:09 +0000 (11:32 +0000)]
Remove unused ExperimentalNS API in AudioProcessing

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

Review URL: https://webrtc-codereview.appspot.com/12909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6718 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAudioBuffer: Eliminate the SplitChannelBuffer class
kwiberg@webrtc.org [Thu, 17 Jul 2014 09:46:37 +0000 (09:46 +0000)]
AudioBuffer: Eliminate the SplitChannelBuffer class

It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.

R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMove additional state into WebRtcVideoSendStream.
pbos@webrtc.org [Thu, 17 Jul 2014 08:51:46 +0000 (08:51 +0000)]
Move additional state into WebRtcVideoSendStream.

Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoSimplify AudioBuffer::mixed_low_pass_data API
aluebs@webrtc.org [Thu, 17 Jul 2014 08:27:39 +0000 (08:27 +0000)]
Simplify AudioBuffer::mixed_low_pass_data API

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

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
kwiberg@webrtc.org [Thu, 17 Jul 2014 08:18:33 +0000 (08:18 +0000)]
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter

R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd unit test for MediaFile WAV file writing
kwiberg@webrtc.org [Thu, 17 Jul 2014 08:11:32 +0000 (08:11 +0000)]
Add unit test for MediaFile WAV file writing

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6713 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoFixes up rtc so that it compiles on iOS 8 SDK.
tkchin@webrtc.org [Thu, 17 Jul 2014 00:21:59 +0000 (00:21 +0000)]
Fixes up rtc so that it compiles on iOS 8 SDK.
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as
UIInterfaceOrientationPortrait.

R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13029004

Patch from David Maclachlan <dmaclach@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6712 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert 6707 "Add support of multiple STUN servers in UDPPort."
wu@webrtc.org [Thu, 17 Jul 2014 00:03:24 +0000 (00:03 +0000)]
Revert 6707 "Add support of multiple STUN servers in UDPPort."

Reason:
Breaks the build on callclient.cc.

> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
>
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
>
> BUG=3310
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13879004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agor6709 lacks a change in BUILD.gn
minyue@webrtc.org [Wed, 16 Jul 2014 22:18:49 +0000 (22:18 +0000)]
r6709 lacks a change in BUILD.gn

BUG=
R=marpan@google.com, marpan@webrtc.org, pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6710 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRaw packet loss rate reported by RTP_RTCP module may vary too drastically over time...
minyue@webrtc.org [Wed, 16 Jul 2014 21:28:26 +0000 (21:28 +0000)]
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.

The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

BUG=
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMake sure b lines appear before all the a lines. Per RFC 4566, the order of media...
wu@webrtc.org [Wed, 16 Jul 2014 21:03:13 +0000 (21:03 +0000)]
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
  m=  (media name and transport address)
  i=* (media title)
  c=* (connection information -- optional if included at
       session level)
  b=* (zero or more bandwidth information lines)
  k=* (encryption key)
  a=* (zero or more media attribute lines)

BUG=2260
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd support of multiple STUN servers in UDPPort.
jiayl@webrtc.org [Wed, 16 Jul 2014 20:55:31 +0000 (20:55 +0000)]
Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.

I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.

BUG=3310
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoCompile-time guard for iOS7 specific property.
tkchin@webrtc.org [Wed, 16 Jul 2014 19:59:05 +0000 (19:59 +0000)]
Compile-time guard for iOS7 specific property.

BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days ago(Auto)update libjingle 71240799-> 71250251
buildbot@webrtc.org [Wed, 16 Jul 2014 14:23:08 +0000 (14:23 +0000)]
(Auto)update libjingle 71240799-> 71250251

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoPrint an info log instead of return an error if an external encoder is de-registered...
stefan@webrtc.org [Wed, 16 Jul 2014 11:20:40 +0000 (11:20 +0000)]
Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.

This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6704 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemove old padding path in RTPSender.
pbos@webrtc.org [Wed, 16 Jul 2014 09:37:29 +0000 (09:37 +0000)]
Remove old padding path in RTPSender.

Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoint16<->float conversions: Use size_t for array length argument, not int
kwiberg@webrtc.org [Wed, 16 Jul 2014 08:36:52 +0000 (08:36 +0000)]
int16<->float conversions: Use size_t for array length argument, not int

size_t is more appropriate for array lengths, since int might
theoretically be too small for a really large array. But more
importantly, if the caller's value is naturally of type size_t and the
function requires an int, VC++ will trigger warning C4267
(http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the
implicit cast might be lossy, forcing the caller to do a manual cast.
Typing the function with size_t in the first place resolves the
problem.

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoDefine convenient FATAL_ERROR() and FATAL_ERROR_IF() macros
kwiberg@webrtc.org [Wed, 16 Jul 2014 08:34:58 +0000 (08:34 +0000)]
Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6701 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agonrsh1 is written before tmp321 is read, so needs to be earlyclobber
kwiberg@webrtc.org [Wed, 16 Jul 2014 08:26:48 +0000 (08:26 +0000)]
nrsh1 is written before tmp321 is read, so needs to be earlyclobber

Otherwise, the compiler is allowed to put them in the same register
under the assumption that all inputs are read before any
(non-earlyclobber) output is written, which in this case would result
in nrsh2 being corrupted.

BUG=3439
R=aluebs@webrtc.org, ljubomir.papuga@gmail.com

Review URL: https://webrtc-codereview.appspot.com/16089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6700 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoImplement unittest for SetSendCodecsChangesExistingStreams.
pbos@webrtc.org [Wed, 16 Jul 2014 08:01:38 +0000 (08:01 +0000)]
Implement unittest for SetSendCodecsChangesExistingStreams.

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19869004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6699 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix an invalid memory access due to typo in win/cursor.cc.
jiayl@webrtc.org [Tue, 15 Jul 2014 20:32:03 +0000 (20:32 +0000)]
Fix an invalid memory access due to typo in win/cursor.cc.

BUG=crbug/391468
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoAfter an audio interruption the audio unit no longer invokes its render callback...
tkchin@webrtc.org [Tue, 15 Jul 2014 20:20:47 +0000 (20:20 +0000)]
After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.

CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoMinor refactoring of StatsCollector.
tommi@webrtc.org [Tue, 15 Jul 2014 19:22:37 +0000 (19:22 +0000)]
Minor refactoring of StatsCollector.
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.

The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemove Thread::RunningForChannelManager().
tkchin@webrtc.org [Tue, 15 Jul 2014 17:52:43 +0000 (17:52 +0000)]
Remove Thread::RunningForChannelManager().

I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoImprovements to the pacer where it lost some budget due to truncation errors.
stefan@webrtc.org [Tue, 15 Jul 2014 16:40:38 +0000 (16:40 +0000)]
Improvements to the pacer where it lost some budget due to truncation errors.

With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.

We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.

BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix breakage introduced by r6691.
pbos@webrtc.org [Tue, 15 Jul 2014 15:51:33 +0000 (15:51 +0000)]
Fix breakage introduced by r6691.

ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.

BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMake RTCP sender report send media bytes.
pbos@webrtc.org [Tue, 15 Jul 2014 15:25:39 +0000 (15:25 +0000)]
Make RTCP sender report send media bytes.

r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoEliminate unnecessary #include
kwiberg@webrtc.org [Tue, 15 Jul 2014 12:50:13 +0000 (12:50 +0000)]
Eliminate unnecessary #include

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6690 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agortc::Fatal output: Print space between # and message
kwiberg@webrtc.org [Tue, 15 Jul 2014 11:41:05 +0000 (11:41 +0000)]
rtc::Fatal output: Print space between # and message

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6689 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRemove the VPM denoiser.
pbos@webrtc.org [Tue, 15 Jul 2014 09:50:40 +0000 (09:50 +0000)]
Remove the VPM denoiser.

The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoHandle the case if an unusually long peer name is provided in the peerconnection...
tommi@webrtc.org [Tue, 15 Jul 2014 08:56:07 +0000 (08:56 +0000)]
Handle the case if an unusually long peer name is provided in the peerconnection example.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoReplace strcpy with talk_base::strcpyn.
pbos@webrtc.org [Tue, 15 Jul 2014 08:28:20 +0000 (08:28 +0000)]
Replace strcpy with talk_base::strcpyn.

Cpplint reports error 'Almost always, snprintf is better than strcpy'
when checking code styles. The function talk_base::strcpyn() is a better
option than strcpy().

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12919004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6686 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRoll libyuv from 1033 to 1035 to get cpuid fix for AVX2 that avoids misdetect causing...
fbarchard@google.com [Mon, 14 Jul 2014 23:27:05 +0000 (23:27 +0000)]
Roll libyuv from 1033 to 1035 to get cpuid fix for AVX2 that avoids misdetect causing a crash in AVX2 code on cpus that do not have AVX2.
BUG=libyuv:343
TESTED=libyuv try bots pass
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6685 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRoll chromium 282462:282879.
fgalligan@google.com [Mon, 14 Jul 2014 23:14:48 +0000 (23:14 +0000)]
Roll chromium 282462:282879.

Pick up the libvpx roll:
https://codereview.chromium.org/387003005/

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6684 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRebase webrtc/base with r6682 version of talk/base:
henrike@webrtc.org [Mon, 14 Jul 2014 22:03:57 +0000 (22:03 +0000)]
Rebase webrtc/base with r6682 version of talk/base:
cls ported: r6671, r6672, r6679 (reverts and unreverts in r6680, r6682).
svn diff -r 6656:6682 http://webrtc.googlecode.com/svn/trunk/talk/base >
6682.diff
sed -i.bak "s/talk_base/rtc/g" 6682.diff
sed -i.bak "s/#ifdef WIN32/#if defined(WEBRTC_WIN)/g" 6682.diff
sed -i.bak "s/#if defined(WIN32)/#if defined(WEBRTC_WIN)/g" 6682.diff
patch -p0 -i 6682.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6683 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdd a facility to the Thread class to catch blocking regressions.
henrike@webrtc.org [Mon, 14 Jul 2014 21:42:39 +0000 (21:42 +0000)]
Add a facility to the Thread class to catch blocking regressions.

This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.

This is a reland of an already reviewed cl (r6679) that got reverted by mistake.

TBR=xians@google.com,tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6682 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoEnable SCTP compile for iOS.
tkchin@webrtc.org [Mon, 14 Jul 2014 20:24:09 +0000 (20:24 +0000)]
Enable SCTP compile for iOS.

Chromium's been updated to pull a version of usrsctplib that will compile correctly. This update DEPS to point at new revision and turn on the compile time flags for iOS sctp.

BUG=3211
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6681 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71116846-> 71117224
buildbot@webrtc.org [Mon, 14 Jul 2014 20:22:21 +0000 (20:22 +0000)]
(Auto)update libjingle 71116846-> 71117224

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6680 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdd a facility to the Thread class to catch blocking regressions.
tommi@webrtc.org [Mon, 14 Jul 2014 20:21:36 +0000 (20:21 +0000)]
Add a facility to the Thread class to catch blocking regressions.

This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6679 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoA step towards changing StatsReport::Value::name to an enum.
tommi@webrtc.org [Mon, 14 Jul 2014 20:19:56 +0000 (20:19 +0000)]
A step towards changing StatsReport::Value::name to an enum.
The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMake StatsCollector depend on always having a valid session pointer.
tommi@webrtc.org [Mon, 14 Jul 2014 20:15:26 +0000 (20:15 +0000)]
Make StatsCollector depend on always having a valid session pointer.
This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/13959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMinor refactoring of the session classes.
tommi@webrtc.org [Mon, 14 Jul 2014 20:11:49 +0000 (20:11 +0000)]
Minor refactoring of the session classes.
Make member variables that never change and are touched on multiple threads, const.
Move implementations of setters/getters of variables that can change, into the cc file in preparation of adding thread correctness checks.

This is a relanding of a cl already reviewed but got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6676 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71107853-> 71115715
buildbot@webrtc.org [Mon, 14 Jul 2014 20:05:09 +0000 (20:05 +0000)]
(Auto)update libjingle 71107853-> 71115715

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71099685-> 71107853
buildbot@webrtc.org [Mon, 14 Jul 2014 18:22:37 +0000 (18:22 +0000)]
(Auto)update libjingle 71099685-> 71107853

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6674 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoFix deadlock in Android stopCapture() call.
glaznev@webrtc.org [Mon, 14 Jul 2014 17:01:53 +0000 (17:01 +0000)]
Fix deadlock in Android stopCapture() call.

BUG=3467
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6673 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoFix a type cast issue for compiling webrtc with BoringSSL.
jiayl@webrtc.org [Mon, 14 Jul 2014 16:42:46 +0000 (16:42 +0000)]
Fix a type cast issue for compiling webrtc  with BoringSSL.

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

Review URL: https://webrtc-codereview.appspot.com/16039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6672 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 70948025-> 70959275
buildbot@webrtc.org [Mon, 14 Jul 2014 14:54:16 +0000 (14:54 +0000)]
(Auto)update libjingle 70948025-> 70959275

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6671 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoGN: Fix include paths for WebRTC in Chromium build.
kjellander@webrtc.org [Sun, 13 Jul 2014 09:02:54 +0000 (09:02 +0000)]
GN: Fix include paths for WebRTC in Chromium build.

Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.

This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.

However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.

BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoFix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
tommi@webrtc.org [Fri, 11 Jul 2014 20:33:39 +0000 (20:33 +0000)]
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .

A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.

TBR=pbos,stefan

Review URL: https://webrtc-codereview.appspot.com/13939005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemove always-true expression.
tommi@webrtc.org [Fri, 11 Jul 2014 19:34:54 +0000 (19:34 +0000)]
Remove always-true expression.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/16059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoLanding pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
tommi@webrtc.org [Fri, 11 Jul 2014 19:09:59 +0000 (19:09 +0000)]
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
---

Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition

This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional

This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).

BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoThread annotate RTCPSender.
pbos@webrtc.org [Fri, 11 Jul 2014 15:36:26 +0000 (15:36 +0000)]
Thread annotate RTCPSender.

Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoFixing memcheck leak suppressions for XMPPClient tests.
pbos@webrtc.org [Fri, 11 Jul 2014 13:44:45 +0000 (13:44 +0000)]
Fixing memcheck leak suppressions for XMPPClient tests.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6665 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMove pacer to fully use webrtc::Clock instead of webrtc::TickTime.
stefan@webrtc.org [Fri, 11 Jul 2014 13:44:02 +0000 (13:44 +0000)]
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoImplement unittest SetRecvCodecsAcceptDefaultCodecs.
pbos@webrtc.org [Fri, 11 Jul 2014 13:02:54 +0000 (13:02 +0000)]
Implement unittest SetRecvCodecsAcceptDefaultCodecs.

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14869004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6663 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoCast payload types to int for logging.
pbos@webrtc.org [Fri, 11 Jul 2014 12:33:45 +0000 (12:33 +0000)]
Cast payload types to int for logging.

uint8_t gets interpreted as char and printed as such, instead of being
printed in decimal, casting them to int allows us to read what payload
types are actually used without converting them from ASCII first.

BUG=chromium:390874
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6662 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoDocument that channels are stored contiguously in AudioBuffer
aluebs@webrtc.org [Fri, 11 Jul 2014 11:40:48 +0000 (11:40 +0000)]
Document that channels are stored contiguously in AudioBuffer

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

Review URL: https://webrtc-codereview.appspot.com/20909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6661 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRemove unnecessary build message.
tommi@webrtc.org [Fri, 11 Jul 2014 11:15:35 +0000 (11:15 +0000)]
Remove unnecessary build message.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRemove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org [Fri, 11 Jul 2014 09:55:30 +0000 (09:55 +0000)]
Remove the send-side cname getter APIs from voice and video engine.

These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRoll chromium_revision 280876:282462
henrikg@webrtc.org [Fri, 11 Jul 2014 08:10:19 +0000 (08:10 +0000)]
Roll chromium_revision 280876:282462

No significant DEPS changes in this roll, only some changes in how clang_format is downloaded.

clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.

R=henrika@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6658 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoroll libyuv to r1033 for clang-cl support on windows.
fbarchard@google.com [Thu, 10 Jul 2014 23:40:15 +0000 (23:40 +0000)]
roll libyuv to r1033 for clang-cl support on windows.
BUG=chromium:391927
TESTED=manual testing libyuv compiles with clang-cl
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6657 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRebase webrtc/base with r6655 version of talk/base:
henrike@webrtc.org [Thu, 10 Jul 2014 22:47:02 +0000 (22:47 +0000)]
Rebase webrtc/base with r6655 version of talk/base:
cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)):
svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff
sed -i.bak "s/talk_base/rtc/g" 6655.diff
patch -p0 -i 6555.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6656 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoCount total bytes sent in RTPSender::Bytes().
pbos@webrtc.org [Thu, 10 Jul 2014 16:24:54 +0000 (16:24 +0000)]
Count total bytes sent in RTPSender::Bytes().

Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d