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Author SHA1 Message Date
henrik.lundin@webrtc.org
3a5825909d Deleting all ACM1 files 
ACM1 is deprecated and replaced by ACM2
(webrtc/modules/audio_coding/acm2/).

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 08:08:56 +00:00
stefan@webrtc.org
46e636a3f5 Fix failing test introduced with r6111. 
Test was assuming that getting the receive estimate of a stream which hasn't received packets would return an error, new behavior is to return 0.

TBR=wu@webrtc.org
BUG=crbug/371714

Review URL: https://webrtc-codereview.appspot.com/21419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:17:29 +00:00
buildbot@webrtc.org
eaf2bd916b (Auto)update libjingle 66813165-> 66836233 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:12:19 +00:00
mallinath@webrtc.org
d37bcfa882 Changed enums to less generic names. 
IPv4/IPv6 will be sent when RegisterUMAObserver is called. This is done
as Initialize is not called through interface.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:10:18 +00:00
stefan@webrtc.org
72885d1c91 Fixes log spam introduced with r6041. 
We shouldn't return an error if we don't yet have a valid estimate.

BUG=crbug/371714
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 22:09:27 +00:00
buildbot@webrtc.org
17911dca80 (Auto)update libjingle 66798415-> 66813165 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:42:49 +00:00
henrike@webrtc.org
0df2ea064f Rollback of r6108 
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:41:12 +00:00
pbos@webrtc.org
a7f70a487f Initialize bitrates in ValidateCodecFormat. 
Attempt to un-break a Visual Studio build (unknown version) that
incorrectly reports that these are potentially uninitialized.

BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:20:40 +00:00
henrike@webrtc.org
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. 
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:03:09 +00:00
henrike@webrtc.org
f3a5e6afc4 Suppression for WebRtcVideoChannel2BaseTest.SetSendSsrc. 
BUG=3336
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 17:58:21 +00:00
henrike@webrtc.org
d886e4aaf7 Suppression for test failing on dr memory (in waterfall). 
BUG=3336
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 16:31:21 +00:00
pbos@webrtc.org
d266a2020f Initial wiring of new webrtc API in libjingle. 
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 14:32:01 +00:00
henrika@webrtc.org
6b02eea6ac Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:24:10 +00:00
henrika@webrtc.org
1cec3957b8 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:19:19 +00:00
kwiberg@webrtc.org
924e81f797 Echo cancellation functions docs: Follow style guide w.r.t. placement of * 
The style guide says to use "void* x", not void *x", and the code in
these files already do so, but the comments do not. Fix that.

Also, in the interest of reducing eye strain, I fixed the vertical
alignment in a small number of cases.

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 09:55:19 +00:00
henrika@webrtc.org
66021e0fa2 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 08:53:27 +00:00
turaj@webrtc.org
b9863ce6ba One of the NetEq methods needs to be virtual. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-10 00:58:49 +00:00
turaj@webrtc.org
e14ffaa40b Update DEPS to pull r6096 changes to third_party/openmax_dl/dl/dl.gyp 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 21:40:23 +00:00
mallinath@webrtc.org
0f2a22b3fa Removed sending metrics from PeerConnection about IPv4 and IPv6. 
Reasons: 1: There is memcheck failure.
         2: DoInitialize is called before RegisterUMAObserver,
            which means this will be never triggered in real cases.

BUG=3326
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 21:15:06 +00:00
buildbot@webrtc.org
8a54844333 (Auto)update libjingle 66624678-> 66643715 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 18:10:55 +00:00
turaj@webrtc.org
17bf9a2c5e Modifying neteq.gyp 
|codecs| list is introduced, which is used in both |neteq_dependencies| and AudiDecoder unittests dependencies. This way, AudioDecoder unittests depend only on |codecs| and not on the entire |neteq_dependencies|, which is unnecessary.

TEST=trybots
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 18:04:50 +00:00
buildbot@webrtc.org
1cd14a4502 (Auto)update libjingle 66556498-> 66624678 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 15:01:40 +00:00
henrika@webrtc.org
3b76627afe Removes parts of the webrtc::VoEHardware sub API (relanding) 
Relanding https://webrtc-codereview.appspot.com/18399004/

TBR=niklase

Review URL: https://webrtc-codereview.appspot.com/16489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:43:00 +00:00
henrika@webrtc.org
3106b706c0 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." 
> Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
> 
> BUG=3206
> R=andrew@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18399004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:10:50 +00:00
henrika@webrtc.org
9de3d844ae Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 10:55:11 +00:00
andresp@webrtc.org
6a8a6723d3 FieldTrial implementation for webrtc. 
BUG=crbug/367114
R=asvitkine@chromium.org, mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 07:14:34 +00:00
buildbot@webrtc.org
ca27236272 (Auto)update libjingle 66541346-> 66556498 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 23:10:23 +00:00
wu@webrtc.org
02b286bfc9 Raise kViEMaxNumberOfChannels from 32 to 64 
Recent testing has shown that on modern desktops and laptops, decoding more than
32 low-resolution realtime video streams simultaneously is both possible and
desirable.

Reviewed:
https://webrtc-codereview.appspot.com/16449004/

TBR=mflodman
BUG=

Review URL: https://webrtc-codereview.appspot.com/17429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 22:22:41 +00:00
buildbot@webrtc.org
1567b8cf8c (Auto)update libjingle 66540208-> 66541346 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:54:16 +00:00
buildbot@webrtc.org
073dfdd10a (Auto)update libjingle 66539128-> 66540208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:36:21 +00:00
buildbot@webrtc.org
d1ae89fae1 (Auto)update libjingle 66524760-> 66539128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:19:26 +00:00
elham@webrtc.org
e37951d28f Updated WebRTC version to 3.53 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 17:09:31 +00:00
buildbot@webrtc.org
ff6a3d920a (Auto)update libjingle 66523887-> 66524760 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:16:41 +00:00
jiayl@webrtc.org
f7026cd7c8 Check SCTP_EWOULDBLOCK instead of EWOULDBLOCK in SctpDataMediaChannel. 
usrsctp.h redefines EWOULDBLOCK to WSAEWOULDBLOCK on Windows, but usrsctp_sendv still returns the BSD EWOULDBLOCK (i.e. SCTP_EWOURLBLOCK) when sending data fails due to congestion.
We will need to revert this change when usersctp is fixed.

BUG=2866
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:02:23 +00:00
buildbot@webrtc.org
c5bb22395c (Auto)update libjingle 66424806-> 66523513 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:00:58 +00:00
kjellander@webrtc.org
9e230eae82 DrMemory: Removing suppression as Dr Memory was fixed. 
According to
https://code.google.com/p/webrtc/issues/detail?id=3275
the issue is now fixed in the drmemory.DEPS of r267732.
Since we don't roll this DEPS (it's automatically updated
as it's a separate solution in the checkout for these bots)
we already have this update.

BUG=3275
TEST=Passing trybot: git try --bot=win_drmemory_light
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 12:24:17 +00:00
kwiberg@webrtc.org
4cc763621e AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ 
data_was_mixed_ was always false, so it can be removed. That makes the
role of data_ simpler, but not so simple that it doesn't merit an
explanation.

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 07:10:11 +00:00
buildbot@webrtc.org
2219037e5e (Auto)update libjingle 66406192-> 66424806 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 17:52:33 +00:00
wu@webrtc.org
66773a032a Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. 
BUG=3111
TEST=try bots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 17:09:44 +00:00
henrike@webrtc.org
25a344edc6 WebRtcVideoEngineTestFake.SendReceiveBitratesStats suppressed for "Win DrMemory Full" 
BUG=11288120
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 16:41:02 +00:00
buildbot@webrtc.org
dd4742a9ef (Auto)update libjingle 66388864-> 66406192 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 14:50:35 +00:00
buildbot@webrtc.org
ed97bb0eb4 (Auto)update libjingle 66340694-> 66388864 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 11:15:20 +00:00
braveyao@webrtc.org
94f1d4cd55 Fix odd codes in video_capture on Mac. 
BUG=3272
TEST=vie_auto_test
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 02:57:13 +00:00
buildbot@webrtc.org
f9277a9381 (Auto)update libjingle 66326258-> 66340694 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 00:29:05 +00:00
fischman@webrtc.org
b1eb43142e video_render.gypi: clean up some libraries directives to be more specific. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 00:09:30 +00:00
buildbot@webrtc.org
861d4b0de9 (Auto)update libjingle 66322380-> 66326258 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 22:11:02 +00:00
henrike@webrtc.org
3129e684a3 openmax_dl was not added to .gitignore in r6037. 
BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:47:45 +00:00
buildbot@webrtc.org
0581f0ba0a (Auto)update libjingle 66303009-> 66322380 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:36:31 +00:00
buildbot@webrtc.org
a18b4c96af (Auto)update libjingle 66301332-> 66303009 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:48:14 +00:00
buildbot@webrtc.org
e65c9a6e67 (Auto)update libjingle 66299810-> 66301332 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:28:28 +00:00