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Author SHA1 Message Date
buildbot@webrtc.org
e2da234e27 (Auto)update libjingle 71766184-> 71775619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:09:01 +00:00
buildbot@webrtc.org
21b4da8ebd (Auto)update libjingle 71753329-> 71766184 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 19:07:53 +00:00
tommi@webrtc.org
0f7328cd6b 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
 2014-07-23 16:31:57 +00:00
pbos@webrtc.org
9359cb3e75 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
 2014-07-23 15:44:48 +00:00
stefan@webrtc.org
f24c4a3b8d 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
 2014-07-23 10:27:41 +00:00
pbos@webrtc.org
5ff71ab4b3 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
 2014-07-23 07:28:56 +00:00
buildbot@webrtc.org
89c833cd9d (Auto)update libjingle 71726409-> 71726772 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:11:58 +00:00
buildbot@webrtc.org
f67f6aa741 (Auto)update libjingle 71675033-> 71726409 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:04:22 +00:00
pbos@webrtc.org
8120353342 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
 2014-07-23 07:04:08 +00:00
pbos@webrtc.org
543e589205 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
 2014-07-23 07:01:31 +00:00
glaznev@webrtc.org
efe4b9af49 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
 2014-07-22 17:44:53 +00:00
pbos@webrtc.org
6f48f1bf68 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
 2014-07-22 16:29:54 +00:00
pbos@webrtc.org
cadd078cf9 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
 2014-07-22 15:26:09 +00:00
minyue@webrtc.org
194fea7640 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
 2014-07-22 09:55:51 +00:00
pbos@webrtc.org
85f42949d6 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
 2014-07-22 09:14:58 +00:00
andresp@webrtc.org
b0c8228755 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
 2014-07-22 07:17:17 +00:00
andresp@webrtc.org
5ab7616983 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
 2014-07-22 06:48:58 +00:00
buildbot@webrtc.org
fa5fcd671d (Auto)update libjingle 71599033-> 71605904 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 21:55:43 +00:00
buildbot@webrtc.org
e69b061926 (Auto)update libjingle 71575585-> 71599033 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 20:38:58 +00:00
andrew@webrtc.org
ceafa8cce9 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
 2014-07-21 16:43:13 +00:00
tommi@webrtc.org
908f57ed94 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
 2014-07-21 11:44:39 +00:00
tommi@webrtc.org
756b8462eb 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
 2014-07-21 11:24:17 +00:00
tommi@webrtc.org
fd61a1d693 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
 2014-07-21 11:05:28 +00:00
tommi@webrtc.org
647e05cfcd 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
 2014-07-21 10:55:11 +00:00
pbos@webrtc.org
3c10758b3b 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
 2014-07-20 15:27:35 +00:00
pbos@webrtc.org
8fdeee6abf 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
 2014-07-20 14:40:23 +00:00
jiayl@webrtc.org
3edbaaf337 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
 2014-07-18 23:57:50 +00:00
buildbot@webrtc.org
99f6308a2d (Auto)update libjingle 71460499-> 71464449 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 23:31:30 +00:00
jiayl@webrtc.org
a0b929b63c 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
 2014-07-18 22:28:36 +00:00
buildbot@webrtc.org
196ae6d667 (Auto)update libjingle 71456344-> 71456420 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:41:41 +00:00
buildbot@webrtc.org
3dec81a736 (Auto)update libjingle 71456173-> 71456344 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:39:56 +00:00
jiayl@webrtc.org
a6e8cf8fb7 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
 2014-07-18 21:34:11 +00:00
minyue@webrtc.org
f563e85ab0 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
 2014-07-18 21:11:27 +00:00
buildbot@webrtc.org
60e65b11c1 (Auto)update libjingle 71452608-> 71453580 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:07:50 +00:00
jiayl@webrtc.org
8636fc852e 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
 2014-07-18 20:54:27 +00:00
tkchin@webrtc.org
ff50debd37 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
 2014-07-18 17:17:59 +00:00
tkchin@webrtc.org
9343cf67a9 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
 2014-07-18 17:13:28 +00:00
pbos@webrtc.org
ba92c52570 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
 2014-07-18 13:33:48 +00:00
minyue@webrtc.org
026859b983 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
 2014-07-18 12:28:28 +00:00
pbos@webrtc.org
e6f84ae8a6 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
 2014-07-18 11:11:55 +00:00
pbos@webrtc.org
e9e4253a3c 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
 2014-07-18 10:12:50 +00:00
pbos@webrtc.org
d1ea06b3d5 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
 2014-07-18 09:35:58 +00:00
buildbot@webrtc.org
c31651d847 (Auto)update libjingle 71378257-> 71410012 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 08:22:39 +00:00
kwiberg@webrtc.org
e364ac902f 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
 2014-07-18 07:50:29 +00:00
andrew@webrtc.org
c145668dc8 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
 2014-07-17 23:16:44 +00:00
wu@webrtc.org
4f5da030f1 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
 2014-07-17 22:19:21 +00:00
mallinath@webrtc.org
aa93611375 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
 2014-07-17 21:55:04 +00:00
mallinath@webrtc.org
e5995aadd5 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
 2014-07-17 18:23:52 +00:00
jiayl@webrtc.org
e10d28cf14 fix 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 17:07:49 +00:00
stefan@webrtc.org
8b94e3da0f 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
 2014-07-17 16:10:14 +00:00