wu@webrtc.org
822fbd8b68
Update talk to 50918584.
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Together with Stefan's http://review.webrtc.org/1960004/ .
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
tnakamura@webrtc.org
aa4d96a134
Revert r4301
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R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
stefan@webrtc.org
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
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rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
solenberg@webrtc.org
a6db54d4c9
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup.
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- Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1553005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 16:02:56 +00:00
stefan@webrtc.org
c74c3c2447
Adds integration test for RTX and fixes bugs found.
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BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 13:48:22 +00:00
solenberg@webrtc.org
cb9cff0c71
Add functions to ViE API to enable/disable the absolute send time header extension.
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BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1487004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4065 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-20 12:00:23 +00:00
pbos@webrtc.org
f5d4cb1958
Include files from webrtc/.. paths in video_engine/
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BUG=1662
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1492004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 13:44:48 +00:00
mflodman@webrtc.org
4dee30927a
Remove SetOverUseDetectorOptions and cleaned ViESharedData.
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R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1486004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:13:18 +00:00
andresp@webrtc.org
29b2219914
Adding a factory to remote bitrate estimator and allow it to be set via config.
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Additionally:
- clean api to set remote bitrate estimator mode.
- clean api to set over use detector options.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1448006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:10:58 +00:00
andrew@webrtc.org
d72262dc01
Fix compile errors in ViE with latest clang.
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Rolling to the latest Chromium picks up a new clang, which catches a fresh error:
error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~
static_cast
This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).
Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
AutoTestSleep(std::numeric_limits<long>::max());
~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This fixes the errors and is required before stable can be rolled in Chromium.
TBR=mflodman,andresp
Review URL: https://webrtc-codereview.appspot.com/1450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 02:12:07 +00:00
stefan@webrtc.org
8ca8a71de2
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
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This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.
BUG=1613
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1327008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 16:48:32 +00:00
stefan@webrtc.org
ccd4b2aec8
Add a default RTT to CallStats and use different values for buffered/real-time mode.
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BUG=1613
Review URL: https://webrtc-codereview.appspot.com/1326007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 15:58:23 +00:00
mflodman@webrtc.org
9f5ebb5251
Adding a payload type for RTX.
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BUG=736
TEST=Modified RTP unittests.
Review URL: https://webrtc-codereview.appspot.com/1278004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-12 14:55:46 +00:00
pbos@webrtc.org
b238d1210b
WebRtc_Word32 -> int32_t in video_engine/
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BUG=314
Review URL: https://webrtc-codereview.appspot.com/1302005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 13:41:51 +00:00
mikhal@webrtc.org
ef9f76a59d
Adding a receive side API for buffering mode.
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At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
mikhal@webrtc.org
3d305c64b4
Updates to send side streaming mode:
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1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.
Review URL: https://webrtc-codereview.appspot.com/1101007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-10 18:42:55 +00:00
mflodman@webrtc.org
4fd5527ab1
Don't report an error for GetEstimatedReceiveBandwidth if there is no valid
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estimate.
BUG=1377
Review URL: https://webrtc-codereview.appspot.com/1095005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3479 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-06 17:46:39 +00:00
mikhal@webrtc.org
dbe97d2550
Adding a send side API for streaming
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Review URL: https://webrtc-codereview.appspot.com/1070009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 19:33:21 +00:00
mflodman@webrtc.org
d6ec386ff5
Revert the revert in r2988 since that wasn't the issue.
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Review URL: https://webrtc-codereview.appspot.com/931005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25 11:30:29 +00:00
vikasmarwaha@webrtc.org
8239ca5096
Reverse Merged r2884 & r2888 from trunk.
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Review URL: https://webrtc-codereview.appspot.com/929005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 22:35:52 +00:00
andrew@webrtc.org
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00