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Author SHA1 Message Date
henrike@webrtc.org
065247b5b7 Rebase webrtc/base with r6863 version of talk/base: 
cls integrated: r6809
svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff
patch -p0 -i 6809.diff

BUG=3379
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:32:13 +00:00
tommi@webrtc.org
730bf30da7 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 r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:08:33 +00:00
henrik.lundin@webrtc.org
1c8391205e Use test::Packet test::PacketSource classes in neteq_rtpplay 
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:29:38 +00:00
bjornv@webrtc.org
96d8b0e69f Revert 6860 "SSE2 version of SubbandCoherence()" 
> SSE2 version of SubbandCoherence()
> 
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
> 
> The output is bit exact.
> 
> R=bjornv@webrtc.org, cd@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18779004
> 
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:09:13 +00:00
bjornv@webrtc.org
0db82f337f SSE2 version of SubbandCoherence() 
The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
reported by audioproc is ~3.3%

The output is bit exact.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18779004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 10:38:31 +00:00
jiayl@webrtc.org
7ec3f9f838 Fix a bug in parsing IceCandidate with IPV6 address. 
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.

BUG=3669
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 23:09:15 +00:00
buildbot@webrtc.org
9eabe5e912 (Auto)update libjingle 72931377-> 72931377 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:48:28 +00:00
mallinath@webrtc.org
2d60c5e8bc Encoding and Decoding of TCP candidates as defined in RFC 6544. 
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204

Review URL: https://webrtc-codereview.appspot.com/21479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:29:20 +00:00
harryjin@google.com
8c01e59424 Allow root build dependencies to be overridden. 
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/22039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 00:08:58 +00:00
buildbot@webrtc.org
53df88c1bc (Auto)update libjingle 72847605-> 72850595 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:46:01 +00:00
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