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Author SHA1 Message Date
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
buildbot@webrtc.org
0b53bd29af (Auto)update libjingle 66294299-> 66299810 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:12:36 +00:00
buildbot@webrtc.org
150835ea34 (Auto)update libjingle 66236292-> 66294299 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 15:54:38 +00:00
tina.legrand@webrtc.org
cfb18dd7a3 Rolling new version of Opus.gyp 
The new version enables optimizations on iOS.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 11:57:01 +00:00
kjellander@webrtc.org
e5e16d7515 Update svn:ignore for resources and third_party. 
This prevents the following from getting wiped on each build:
resources/*.dat
third_party/binutils
third_party/libc++
third_party/libc++abi
third_party/openmax_dl



git-svn-id: http://webrtc.googlecode.com/svn/trunk@6059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 08:14:38 +00:00
wu@webrtc.org
ed4cb56575 Remove timestamp_extrapolator's dependency to Clock and vcm defines. 
TEST=existing tests
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 04:50:49 +00:00
buildbot@webrtc.org
5ee0f05d5f (Auto)update libjingle 66138442-> 66236292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 20:18:08 +00:00
andrew@webrtc.org
382c0c209d Allow the RTP level indicator computation to work at any sample rate. 
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.

We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.

This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:

[ RUN      ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27

BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 18:22:21 +00:00
andrew@webrtc.org
a0edf4cb04 Remove ALLOW_UNUSED. 
Turns out Chromium won't be applying this to COMPILE_ASSERT. We don't
need it at all then.

R=thakis@chromium.org
TBR=thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 18:18:02 +00:00
wu@webrtc.org
0224c20fa6 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. 
* Re-enable test CaptureNtpTimeWithNetworkJitter.
* Use 100ms as the threadhold as a FYI since this is a performance test.

BUG=3271
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 17:42:43 +00:00
jiayl@webrtc.org
4220434d37 Implement the Windows screen capturer using the Magnification API. 
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6048

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 16:08:47 +00:00
tina.legrand@webrtc.org
7dccce3948 Revert 6048 "Implement the Windows screen capturer using the Mag..." 
> Implement the Windows screen capturer using the Magnification API.
> The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
> 
> BUG=2789
> TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
> R=sergeyu@chromium.org, wez@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12149004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 11:17:26 +00:00
braveyao@webrtc.org
633aff6bd0 WebRTCDemo: correct set trace filter operation. 
BUG=3285
TEST=Manul Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 04:24:47 +00:00