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Author SHA1 Message Date
buildbot@webrtc.org
65b98d12c3 (Auto)update libjingle 72839629-> 72847605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:09:08 +00:00
henrike@webrtc.org
3763b9bda0 webrtc/base: removes linkage of crypto 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 21:26:18 +00:00
tkchin@webrtc.org
c8554be6dd Support for TURN/TLS. 
Wrap the socket in an SSL adapter, then simply call StartSSL() on the
SSLAdapter instance.

Cloned from: https://webrtc-codereview.appspot.com/21799004/

R=juberti@chromium.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/14059004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:39:08 +00:00
tkchin@webrtc.org
cb46de24fb Add new OWNERS file to talk/examples. 
R=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:01:34 +00:00
buildbot@webrtc.org
5b1ebacca2 (Auto)update libjingle 72820109-> 72822008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 17:18:00 +00:00
buildbot@webrtc.org
d509678a4e (Auto)update libjingle 72819313-> 72820109 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:57:07 +00:00
buildbot@webrtc.org
94b996cc18 (Auto)update libjingle 72785516-> 72819313 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:47:14 +00:00
stefan@webrtc.org
59a2f9f584 Remove the old H264 code now that a new H.264 packetizer has been implemented. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:09:24 +00:00
stefan@webrtc.org
9d74f7ce8c Fix single nalu packetization bug. 
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:02:16 +00:00
pbos@webrtc.org
e8c84bf4de Fix so video_replay logs aren't spammed. 
Add unknown-SSRC counters instead and log number of unknown packets at
end of session.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/13119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 14:42:45 +00:00
minyue@webrtc.org
1d956dd1a7 Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal. 
It is advisable to set the packet loss rate of FEC conservatively. Say, if the estimated loss rate is 5%, we can set it to 1%. The risk of degradation in quality is small and the overall performance is good.

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:31:36 +00:00
henrik.lundin@webrtc.org
ea25784107 Change how background noise mode in NetEq is set 
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.

In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:27:37 +00:00
buildbot@webrtc.org
476efa2031 (Auto)update libjingle 72785180-> 72785516 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 04:55:21 +00:00
buildbot@webrtc.org
4f0d401fae (Auto)update libjingle 72682155-> 72785180 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 04:47:36 +00:00
harryjin@google.com
aaecefe72a Revert 6839 "Allow root build dependencies to be overridden." 
> Allow root build dependencies to be overridden.
> 
> RISK=P2
> TESTED=manual
> R=andrew@webrtc.org, thorcarpenter@google.com
> 
> Review URL: https://webrtc-codereview.appspot.com/19009004

TBR=harryjin@google.com

Review URL: https://webrtc-codereview.appspot.com/20099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 00:22:57 +00:00
harryjin@google.com
e34abfb8e7 Allow root build dependencies to be overridden. 
RISK=P2
TESTED=manual
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/19009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 23:08:42 +00:00
pbos@webrtc.org
4b5625e5ac RTP video playback tool using Call APIs. 
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 16:26:56 +00:00
stefan@webrtc.org
1ccff349ee Fix crashing fake network pipe tests. 
These tests are not included in bots, this will be fixed in a follow-up by pbos@webrtc.org.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 15:41:58 +00:00
minyue@webrtc.org
2a8df7c375 Fixing two bugs in voe_cmd_test. 
I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 10:05:19 +00:00
stefan@webrtc.org
79c3359e67 Add end-to-end H.264 packetization test. 
Also correctly wires up H.264 packetization in the new Call api.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 09:24:53 +00:00
kjellander@webrtc.org
e415864a32 GN: Add PRESUBMIT.py check for GN changes + default bots. 
Add the GN trybots to the default set and also set them
to be the only bots to run if a CL contains only BUILD.gn
changes.

Update Python exclusions in general and fix a few of the lint
warnings.
The ones in python_charts needs to be disabled since those variables
are actually used when passed via vars() to the template.

BUG=None
TEST=git cl presubmit with the following cases:
A CL with two .gyp changes.
A CL with no changes in .gyp* files.

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

Review URL: https://webrtc-codereview.appspot.com/18719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 09:11:18 +00:00
stefan@webrtc.org
8b033adb19 Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. 
R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 08:06:53 +00:00
jiayl@webrtc.org
56d8e05238 A followup to r6828 to fix a condition check in mediasession.cc. 
BUG=2395
R=juberti@chromium.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:52:36 +00:00
fbarchard@google.com
d7b4dea801 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 
BUG=3660
TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:46:42 +00:00
pbos@webrtc.org
dde16f19e3 Fix some code styles. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22009004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:35:43 +00:00
buildbot@webrtc.org
624a504f5b (Auto)update libjingle 72659510-> 72673987 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 22:13:05 +00:00
jiayl@webrtc.org
e7d47a1473 Maintain the order of the m-lines in CreateOffer and CreateAnswer. 
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.

BUG=2395
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 19:19:05 +00:00
fbarchard@google.com
e086af0fa3 Fix implicite cast from signed int to unsigned int in unittest.cc 
BUG=3636
TESTED=set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Debug
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 17:10:52 +00:00
pbos@webrtc.org
923db6a364 Remove remove_old_gn_binaries DEPS entry. 
Marked for removal at the end of last month.

R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/18049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 16:20:29 +00:00
stefan@webrtc.org
fdcb42dac4 Fix potential crash when depacketizing VP8. 
Caused by a missing check for H264 when reading the RTPVideoTypeHeader union.

R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 13:21:18 +00:00
buildbot@webrtc.org
8e885990ae (Auto)update libjingle 72566057-> 72591796 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 23:56:14 +00:00
henrike@webrtc.org
d6542852f3 Unbreaks linux.cc in Chromium. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 21:51:14 +00:00
jiayl@webrtc.org
b18bf5e47d Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. 
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.

BUG=3282
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 18:34:16 +00:00
fbarchard@google.com
b01ce14b13 add some comments about DEPS lkgr for chromium 
BUG=none
TESTED=none
R=harryjin@google.com

Review URL: https://webrtc-codereview.appspot.com/16209005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 18:07:19 +00:00
henrike@webrtc.org
c9b507253f DrMemory suppression due to r6811. 
BUG=3655
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 16:48:24 +00:00
henrike@webrtc.org
ee135f78b7 Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc. 
BUG=3478
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 15:35:14 +00:00
buildbot@webrtc.org
a27342b7af (Auto)update libjingle 72446860-> 72550257 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 15:22:32 +00:00
minyue@webrtc.org
0040a6ef97 This is a setup to solve 
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 14:41:57 +00:00
asapersson@webrtc.org
84b9e1e9d9 Fix for retransmission. Base layer packets were not retransmitted. 
Issue introduced in r6669.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 11:59:42 +00:00
buildbot@webrtc.org
e0d03f13e4 (Auto)update libjingle 72443101-> 72446860 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 03:04:01 +00:00
buildbot@webrtc.org
6e203d50a3 (Auto)update libjingle 72442050-> 72443101 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 01:13:04 +00:00
buildbot@webrtc.org
52148c2f74 (Auto)update libjingle 72430895-> 72442050 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 00:56:56 +00:00
buildbot@webrtc.org
7cb60ccae1 (Auto)update libjingle 72407428-> 72430895 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 22:03:36 +00:00
buildbot@webrtc.org
3bc48247b7 (Auto)update libjingle 72403605-> 72407428 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 16:53:32 +00:00
buildbot@webrtc.org
6955213eca (Auto)update libjingle 72389720-> 72403605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 15:52:45 +00:00
solenberg@webrtc.org
42d65ce8d7 Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots. 
TBR=hellner
BUG=

Review URL: https://webrtc-codereview.appspot.com/18969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 10:07:46 +00:00
buildbot@webrtc.org
1a678c61f1 (Auto)update libjingle 72320533-> 72380285 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 06:21:50 +00:00
buildbot@webrtc.org
6b21b71068 (Auto)update libjingle 72205295-> 72320533 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 15:08:53 +00:00
stefan@webrtc.org
e1c9caf6ee 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
 2014-07-31 15:07:59 +00:00
stefan@webrtc.org
2ec560606b 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
 2014-07-31 14:59:24 +00:00