buildbot@webrtc.org
|
3e01e0b16c
|
(Auto)update libjingle 66867790-> 66887616
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 17:54:10 +00:00 |
|
henrike@webrtc.org
|
c156174da8
|
Suppressing all tests for WebRtcVideoEngine2 for Win DrMemory Full.
BUG=3336
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6124 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 16:47:32 +00:00 |
|
bjornv@webrtc.org
|
8d63d0ee70
|
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
Rewritten the test to only check for valid volume when we have actually received a value from the audio device. To check if we have actually received a volume value is out of the scope for this test.
BUG=webrtc:367
TESTED=trybots
R=tina.legrand@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6123 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 14:14:56 +00:00 |
|
andresp@webrtc.org
|
93ec9c557b
|
Revert "FieldTrial implementation for webrtc." (rev 6089)
New wiring plans require it to be landed first in chrome for a cleaner roll of webrtc.
BUG=crbug/367114
R=tommi@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6122 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 14:09:40 +00:00 |
|
asapersson@webrtc.org
|
e41dbee8a6
|
Reduced kMaxSampleDiffMs (limit to 22fps).
BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6121 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 13:45:13 +00:00 |
|
pbos@webrtc.org
|
023b101f4e
|
Move gflags usage to video_loopback.
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.
R=mflodman@webrtc.org
BUG=3113
Review URL: https://webrtc-codereview.appspot.com/12379006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 11:26:40 +00:00 |
|
pbos@webrtc.org
|
b5a22b1464
|
Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.
BUG=
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 11:07:01 +00:00 |
|
henrik.lundin@webrtc.org
|
c3e8abda7c
|
Deleting all NetEq3 files
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 10:40:52 +00:00 |
|
henrik.lundin@webrtc.org
|
4d363ae305
|
The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.
This part of the the aftermath of issue 3255.
BUG=3255
R=andrew@webrtc.org, henrike@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6117 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 08:50:02 +00:00 |
|
perkj@webrtc.org
|
e9a604accd
|
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.
http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457
> 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
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-13 08:15:48 +00:00 |
|
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
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2014-05-08 07:10:11 +00:00 |
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buildbot@webrtc.org
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2219037e5e
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(Auto)update libjingle 66406192-> 66424806
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6075 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-05-07 17:52:33 +00:00 |
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wu@webrtc.org
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66773a032a
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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
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2014-05-07 17:09:44 +00:00 |
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henrike@webrtc.org
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25a344edc6
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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
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2014-05-07 16:41:02 +00:00 |
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