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Author SHA1 Message Date
kwiberg@webrtc.org
b3ad8cf6ca Make an AudioEncoder subclass for iSAC 
BUG=3926

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 10:08:19 +00:00
minyue@webrtc.org
abe3f1879c Checking whether ACM uses codec internal or WebRTC DTX. 
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.

This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.

Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.

Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 08:53:21 +00:00
kwiberg@webrtc.org
55d42c32a4 DCHECK: Reference condition parameter in release builds 
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as

  int x = ...
  DCHECK_EQ(x, 17);

R=andrew@webrtc.org

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7858
and reverted: https://code.google.com/p/webrtc/source/detail?r=7859

Review URL: https://webrtc-codereview.appspot.com/31169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 08:32:30 +00:00
phoglund@webrtc.org
cd5b209d68 Deleting quality dashboard code. 
This code has served its purpose and can now be deleted. The dashboard
was mostly defunct, so I've turned it off in AppEngine.

R=kjellander@webrtc.org
BUG=None

Review URL: https://webrtc-codereview.appspot.com/33599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 07:57:22 +00:00
andrew@webrtc.org
3c31e6e2f9 Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon 
WebRtcSpl_MinValueW16Neon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I33c3853766d7594ed121166288e5325a03b3d6fe

Review URL: https://webrtc-codereview.appspot.com/32639004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 00:24:13 +00:00
mflodman@webrtc.org
f4c19480fc Remove jitter_estimate_test.h 
BUG=2156
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 21:08:39 +00:00
aluebs@webrtc.org
c5ebbd98f5 Support 48kHz in Noise Suppression 
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 19:30:57 +00:00
pbos@webrtc.org
d8ca723de7 Remove CELT support from audio_coding. 
R=henrik.lundin@webrtc.org, juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 11:49:13 +00:00
asapersson@webrtc.org
8084f9500f Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. 
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 11:04:13 +00:00
pbos@webrtc.org
85bd53e7c9 Add AbsSendTime unittests to rampup_tests.cc. 
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 10:36:20 +00:00
asapersson@webrtc.org
0df371549f Cast payload type to int in logs. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 10:30:45 +00:00
buildbot@webrtc.org
a85307737c (Auto)update libjingle 81702493-> 81755413 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 09:01:18 +00:00
kwiberg@webrtc.org
3cd26b677a Revert r7858 ("DCHECK: Reference condition parameter in release builds") 
Apparently Visual Studio is cleverer than I am at figuring out what
local variables are actually unused.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 08:57:14 +00:00
kwiberg@webrtc.org
3148060e61 DCHECK: Reference condition parameter in release builds 
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as

  int x = ...
  DCHECK_EQ(x, 17);

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 08:45:47 +00:00
henrik.lundin@webrtc.org
ff1a3e36bd Make an AudioEncoder subclass for comfort noise 
BUG=3926
R=bjornv@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 07:29:08 +00:00
andrew@webrtc.org
6fd52f36db Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon. 
WebRtcSpl_DownsampleFastNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ic43f5452eb7e555b998b1d1e79a9e1530be5c948

Review URL: https://webrtc-codereview.appspot.com/24359004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 00:59:48 +00:00
andrew@webrtc.org
ae20d3bbce Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon. 
WebRtcSpl_CrossCorrelationNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I84f9fb953448b62da452ab8dd60e2c0628293587

Review URL: https://webrtc-codereview.appspot.com/30189004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 23:58:39 +00:00
tommi@webrtc.org
aa2c342c10 Add back a constructor to fix FYI build. 
TBR=perkj

Review URL: https://webrtc-codereview.appspot.com/24349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 20:23:06 +00:00
tommi@webrtc.org
5c32a84620 Attempt to fix FYI bots. 
The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:59:27 +00:00
tkchin@webrtc.org
87776a8935 iAppRTCDemo: WebSocket based signaling. 
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.

BUG=
R=glaznev@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:32:35 +00:00
pthatcher@webrtc.org
0babb4a4e6 Fix a comment. 
R=juberti@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:01:45 +00:00
tommi@webrtc.org
c9d155faeb Move implementation of types in statstypes. to its cc file. 
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer 
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 16:22:09 +00:00
minyue@webrtc.org
19dd129c69 Revert 7846 "Adding DTX to WebRTC Opus wrapper" 
> Adding DTX to WebRTC Opus wrapper
> 
> This is a step toward adding Opus DTX support in WebRTC.
> 
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
> 
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
> 
> We transmit the first 1-byte packet to let decoder be in-sync
> 
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13219004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 15:11:15 +00:00
asapersson@webrtc.org
f244760827 Add histograms for receive statistics: 
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")

BUG=crbug/419657
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 14:13:26 +00:00
minyue@webrtc.org
4321f175f1 Adding DTX to WebRTC Opus wrapper 
This is a step toward adding Opus DTX support in WebRTC.

Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See

https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html

We transmit the first 1-byte packet to let decoder be in-sync

BUG=webrtc:1014
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 13:27:39 +00:00
tommi@webrtc.org
5c3ee4bce6 Add empty implementation file that will hold statstypes.h implementation. 
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:47:01 +00:00
minyue@webrtc.org
1784d7cfad Adding an codec interal CNG test in NetEq. 
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:46:39 +00:00
pbos@webrtc.org
9115cde6c9 Merge VP8 changes. 
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:36:40 +00:00
kwiberg@webrtc.org
e04a93bcf5 Move the AudioDecoder interface out of NetEq 
It belongs with the codecs, next to the AudioEncoder interface.

R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org

Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:12:53 +00:00
asapersson@webrtc.org
97d0489058 Add video send bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")

Add retransmitted bytes to StreamDataCounters.

Change in UpdateRtpStats to also update counters for retransmitted packet.

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 09:47:53 +00:00
kjellander@webrtc.org
7ba9f27f2b Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper. 
This makes it possible to clean up the recipes a lot.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 06:46:13 +00:00
glaznev@webrtc.org
eef85387ec Fix AppRTCDemo closing error for KK and JB Android devices. 
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 01:29:17 +00:00
stefan@webrtc.org
86b6d65ef1 Remove no longer used video codec test framework. 
Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.

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

Review URL: https://webrtc-codereview.appspot.com/31139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 00:02:45 +00:00
henrik.lundin@webrtc.org
8911bc52f1 Add AudioEncoder::Max10MsFramesInAPacket 
BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 21:15:55 +00:00
henrik.lundin@webrtc.org
130fef89dd Bugfix in AudioDecoderTest 
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 21:07:59 +00:00
stefan@webrtc.org
edeea91803 Change all system clock types to int64_t in bitrate_controller. 
They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 19:46:23 +00:00
henrik.lundin@webrtc.org
fcbe36a1d9 Add const qualifier to WebRtcPcm16b_Encode 
BUG=909
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 18:26:49 +00:00
kwiberg@webrtc.org
a1ef7bfa15 ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable. 
Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a
proper definition.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:53:10 +00:00
andrew@webrtc.org
3b3c406908 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." 
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575

> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
> 
> This also add a new build target to build java PeerConnection using Chromes build macros.
> 
> BUG=4031
> R=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28189004

TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:21:50 +00:00
kwiberg@webrtc.org
cb858ba397 Make an AudioEncoder subclass for iLBC 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@google.com
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:11:44 +00:00
bjornv@webrtc.org
ee43263a50 Cleaned up real_fft APIs due to non-existing NEON code 
There are NEON APIs that are not used. Cleaning that up for better overview.

BUG=3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 16:36:22 +00:00
perkj@webrtc.org
ed7824b1c0 Change Android PeerConnectionUnittest to build using Chrome macros. 
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 15:41:01 +00:00
asapersson@webrtc.org
ba8138ba38 Change type of nack_last_time_sent_full_ from uint32_t to int64_t. 
Could cause nack requests to be sent too frequently.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 13:29:02 +00:00
kjellander@webrtc.org
aefe61ae2a PRESUBMIT: Add check for checkdeps. 
Several times I've run into the problem with
presubmit crashing when uploading a CL from a checkout
where gclient sync hasn't run yet.
This will print a user friendly error message instead.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 13:00:30 +00:00
kjellander@webrtc.org
7db359b94a Roll chromium_revision 24b4c73..8e72e1d 
Relevant changes:
* src/buildtools: 6ea835d..535aff2
* src/third_party/android_tools: 4c47ef6..4f723e2
* src/third_party/boringssl/src: 69a0160..00505ec
* src/third_party/icu: 866ff69..53ecf0f
* src/third_party/libvpx: 429874c..9fbec81
Details: 24b4c73..8e72e1d/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 11:48:35 +00:00
kjellander@webrtc.org
d91d359feb PRESUBMIT: Add iOS ARM64 trybots to default set. 
BUG=chromium:436831
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 07:05:38 +00:00
marpan@webrtc.org
fb01376eca Adjust some parameters for VP9 tests. 
Needed for the next/upcoming libvpx roll.

BUG=

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 06:25:51 +00:00
glaznev@webrtc.org
e2a9261f3e Improve AppRTCDemo connection speed by sending all 
http POST requests asynchronously.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 20:11:06 +00:00
kjellander@webrtc.org
bd8cc0b914 Add codereview.settings to the /talk subdirectory 
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 13:47:37 +00:00