external/webrtc.git
4 hours ago(Auto)update libjingle 72205295-> 72320533 master
buildbot@webrtc.org [Thu, 31 Jul 2014 15:08:53 +0000 (15:08 +0000)]
(Auto)update libjingle 72205295-> 72320533

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d

4 hours agoFix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
stefan@webrtc.org [Thu, 31 Jul 2014 15:07:59 +0000 (15:07 +0000)]
Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.

TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d

4 hours agoAdd H.264 packetization.
stefan@webrtc.org [Thu, 31 Jul 2014 14:59:24 +0000 (14:59 +0000)]
Add H.264 packetization.

This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d

7 hours agoAdd simulation of network effects to video_loopback tool.
stefan@webrtc.org [Thu, 31 Jul 2014 12:30:18 +0000 (12:30 +0000)]
Add simulation of network effects to video_loopback tool.

Also add support for uniform random packet loss to the fake network pipe.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d

39 hours agolibjingle: stop building files in talk/base as they are no longer used as of r6799
henrike@webrtc.org [Wed, 30 Jul 2014 04:00:52 +0000 (04:00 +0000)]
libjingle: stop building files in talk/base as they are no longer used as of r6799

BUG=3379
R=thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6802 4adac7df-926f-26a2-2b94-8c16560cd09d

43 hours agoDisable warning 4702 which affects map, xlist and others on vs2012 and vs2013.
fbarchard@google.com [Wed, 30 Jul 2014 00:16:20 +0000 (00:16 +0000)]
Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013.
BUG=3584
TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6801 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoroll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional.
fbarchard@google.com [Tue, 29 Jul 2014 18:07:07 +0000 (18:07 +0000)]
roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional.
BUG=libyuv:346
TESTED=set GYP_DEFINES=target_arch=ia32 libyuv_disable_jpeg=1
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6800 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days ago(Auto)update libjingle 72097588-> 72159069
buildbot@webrtc.org [Tue, 29 Jul 2014 17:36:52 +0000 (17:36 +0000)]
(Auto)update libjingle 72097588-> 72159069

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRemove dependency on openssl for android, add dependency on boringssl. Should make...
solenberg@webrtc.org [Tue, 29 Jul 2014 15:23:59 +0000 (15:23 +0000)]
Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again.

TBR=hellner

Review URL: https://webrtc-codereview.appspot.com/21079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6798 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoUse C functions in aec for MIPS
andrew@webrtc.org [Tue, 29 Jul 2014 14:39:10 +0000 (14:39 +0000)]
Use C functions in aec for MIPS

With GCC 4.9, the MIPS NDK toolchain has been changed to only support 16 spregs by default - the even-numbered ones. This has been changed to support the R6 MIPS architecture. While the old behaviour could be restored by adding "-modd-spreg", this would come with a performance hit because the kernel would emulate odd-numbered spregs and missing R2 instructions.
As a result of this change, the functions removed in this CL no longer compile as there are no longer enough spregs for the compiler to assign. So we are removing these functions and they will use the C implementation until the MIPS code is rewritten.

R=andrew@webrtc.org, ljubomir.papuga@gmail.com, pasko@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16159005

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6797 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoIntegrate rtcp packet class to rtcp receiver tests.
asapersson@webrtc.org [Tue, 29 Jul 2014 08:21:50 +0000 (08:21 +0000)]
Integrate rtcp packet class to rtcp receiver tests.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agomerge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:".
henrike@webrtc.org [Tue, 29 Jul 2014 04:45:23 +0000 (04:45 +0000)]
merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:".

TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/16999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6793 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days ago(Auto)update libjingle 72016417-> 72097588
buildbot@webrtc.org [Mon, 28 Jul 2014 22:26:15 +0000 (22:26 +0000)]
(Auto)update libjingle 72016417-> 72097588

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove a disabled test.
pbos@webrtc.org [Sat, 26 Jul 2014 10:16:49 +0000 (10:16 +0000)]
Remove a disabled test.

ConstrainsSetCodecsAccordingToEncoderConfig has been removed from
webrtcvideoengine_unittest.cc, removing this one as well.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6789 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove clang-format rm_binaries.py DEPS entry.
pbos@webrtc.org [Fri, 25 Jul 2014 23:26:09 +0000 (23:26 +0000)]
Remove clang-format rm_binaries.py DEPS entry.

Breaks runhooks.

BUG=
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6788 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agowebrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed...
henrike@webrtc.org [Fri, 25 Jul 2014 21:58:50 +0000 (21:58 +0000)]
webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325.

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

Review URL: https://webrtc-codereview.appspot.com/21939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6787 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoFix compilation on windows with clang, indentation cleanups
sergeyu@chromium.org [Fri, 25 Jul 2014 19:42:19 +0000 (19:42 +0000)]
Fix compilation on windows with clang, indentation cleanups

R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6779

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoSet NACK/REMB when setting receive codecs.
pbos@webrtc.org [Fri, 25 Jul 2014 19:01:32 +0000 (19:01 +0000)]
Set NACK/REMB when setting receive codecs.

Enabling an additional test to ensure that REMB can be both enabled and
disabled by setting VideoCodecs.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6785 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRoll chromium 282879:285412.
fgalligan@google.com [Fri, 25 Jul 2014 18:58:26 +0000 (18:58 +0000)]
Roll chromium 282879:285412.

Pick up the libvpx roll:
https://codereview.chromium.org/401983003/

R=marpan@google.com
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6784 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRevert of 6778 "Refactor StatsCollector and associated types."
henrike@webrtc.org [Fri, 25 Jul 2014 18:44:42 +0000 (18:44 +0000)]
Revert of 6778 "Refactor StatsCollector and associated types."
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/21969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFixes "argument list too long" problem on Linux by using the "find" command instead...
henrike@webrtc.org [Fri, 25 Jul 2014 18:36:55 +0000 (18:36 +0000)]
Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python.

R=pbos@webrtc.org
TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/18929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6782 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemove timestamp retreival warning/error.
turaj@webrtc.org [Fri, 25 Jul 2014 17:50:10 +0000 (17:50 +0000)]
Remove timestamp retreival warning/error.

An error reported while retreiving playout timestamp if no RTP packet received, yet. This causes an overflow of errors/warnings in applications where few channel are created but only one is actively engaged in a conversation. Therefore, we don't find such logging informative (there is no check upon correctness of timestamp computaion only if a packet already received).

BUG=3545
TEST=manual with voe_cmd_test,try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6781 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRevert "Fix compilation on windows with clang, indentation cleanups"
sergeyu@chromium.org [Fri, 25 Jul 2014 17:37:12 +0000 (17:37 +0000)]
Revert "Fix compilation on windows with clang, indentation cleanups"

This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix compilation on windows with clang, indentation cleanups
sergeyu@chromium.org [Fri, 25 Jul 2014 17:28:25 +0000 (17:28 +0000)]
Fix compilation on windows with clang, indentation cleanups

R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRefactor StatsCollector and associated types.
tommi@webrtc.org [Fri, 25 Jul 2014 10:32:30 +0000 (10:32 +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.

(This is a reland of the original attempt in r6747)

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix a crash in statscollector.cc caused by invoking methods on the worker thread...
jiayl@webrtc.org [Thu, 24 Jul 2014 20:41:20 +0000 (20:41 +0000)]
Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.

BUG=3579
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMake sure padding is sent on the first sending RTP module.
mflodman@webrtc.org [Thu, 24 Jul 2014 16:41:25 +0000 (16:41 +0000)]
Make sure padding is sent on the first sending RTP module.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71829282-> 71834788
buildbot@webrtc.org [Thu, 24 Jul 2014 16:06:35 +0000 (16:06 +0000)]
(Auto)update libjingle 71829282-> 71834788

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRe-revert of 6747 "Refactor StatsCollector and associated types."
henrike@webrtc.org [Thu, 24 Jul 2014 14:20:52 +0000 (14:20 +0000)]
Re-revert of 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/13979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71775619-> 71778545
buildbot@webrtc.org [Wed, 23 Jul 2014 21:40:28 +0000 (21:40 +0000)]
(Auto)update libjingle 71775619-> 71778545

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRevert 6747 "Refactor StatsCollector and associated types."
henrike@webrtc.org [Wed, 23 Jul 2014 21:38:58 +0000 (21:38 +0000)]
Revert 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/18889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRevert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const...
henrike@webrtc.org [Wed, 23 Jul 2014 21:38:09 +0000 (21:38 +0000)]
Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."

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

Review URL: https://webrtc-codereview.appspot.com/18899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 71766184-> 71775619
buildbot@webrtc.org [Wed, 23 Jul 2014 21:09:01 +0000 (21:09 +0000)]
(Auto)update libjingle 71766184-> 71775619

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 71753329-> 71766184
buildbot@webrtc.org [Wed, 23 Jul 2014 19:07:53 +0000 (19:07 +0000)]
(Auto)update libjingle 71753329-> 71766184

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6767 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoTemporarily add a default ctor to StatsReport and make |id| non const.
tommi@webrtc.org [Wed, 23 Jul 2014 16:31:57 +0000 (16:31 +0000)]
Temporarily add a default ctor to StatsReport and make |id| non const.
As soon as I've updated the chrome side, I'll revert this cl.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/16149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6766 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoEnable SendAndReceive tests.
pbos@webrtc.org [Wed, 23 Jul 2014 15:44:48 +0000 (15:44 +0000)]
Enable SendAndReceive tests.

Also fixes a crash in ::SetCapturer which wasn't exposed by tests
before.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6765 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoFix flaky ramp-up test.
stefan@webrtc.org [Wed, 23 Jul 2014 10:27:41 +0000 (10:27 +0000)]
Fix flaky ramp-up test.

Don't require the first estimate to be less than the target bitrate. There are other tests verifying that BWE works, so it's enough for this test to measure the
time it takes to ramp-up.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6764 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoRevert "(Auto)update libjingle 71675033-> 71726409"
pbos@webrtc.org [Wed, 23 Jul 2014 07:28:56 +0000 (07:28 +0000)]
Revert "(Auto)update libjingle 71675033-> 71726409"

This reverts commit r6761 which looks like an accidental auto-revert of
r6760.

BUG=1788
TBR=wu@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6763 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 71726409-> 71726772
buildbot@webrtc.org [Wed, 23 Jul 2014 07:11:58 +0000 (07:11 +0000)]
(Auto)update libjingle 71726409-> 71726772

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6762 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 71675033-> 71726409
buildbot@webrtc.org [Wed, 23 Jul 2014 07:04:22 +0000 (07:04 +0000)]
(Auto)update libjingle 71675033-> 71726409

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6761 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoImplement suspend-below-min-bitrate option.
pbos@webrtc.org [Wed, 23 Jul 2014 07:04:08 +0000 (07:04 +0000)]
Implement suspend-below-min-bitrate option.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6760 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoWire up VideoOptions for payload-based padding.
pbos@webrtc.org [Wed, 23 Jul 2014 07:01:31 +0000 (07:01 +0000)]
Wire up VideoOptions for payload-based padding.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6759 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoAdd VP8 video decoding hw acceleration support to Java Peerconnection library.
glaznev@webrtc.org [Tue, 22 Jul 2014 17:44:53 +0000 (17:44 +0000)]
Add VP8 video decoding hw acceleration support to Java Peerconnection library.
For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.

TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.

BUG=3030
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoImplement encoder options in WebRtcVideoEngine2.
pbos@webrtc.org [Tue, 22 Jul 2014 16:29:54 +0000 (16:29 +0000)]
Implement encoder options in WebRtcVideoEngine2.

Implementing default options to enable denoising by default and wiring
up encoder settings to propagate VP8 settings.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6757 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove unused config.h and math.h includes.
pbos@webrtc.org [Tue, 22 Jul 2014 15:26:09 +0000 (15:26 +0000)]
Remove unused config.h and math.h includes.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6756 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoThe lastest commit on this file was in
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

9 days 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

9 days 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

9 days 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

9 days 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

9 days 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

10 days 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

10 days 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

10 days 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

10 days 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

10 days 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

11 days 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

11 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

12 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

12 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

12 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

12 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

12 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

12 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

12 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

12 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

12 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

13 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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

2 weeks 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