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.
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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
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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
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 01:13:04 +00:00