stefan@webrtc.org
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0a3c1471b8
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Add API to query video engine for the send-side delay.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-05 14:05:07 +00:00 |
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henrik.lundin@webrtc.org
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245037df09
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Remove default implementations for SuspendBelowMinBitrate
These two methods had default implementations while waiting for
changes in libjingle to propagate. Now the changes are in, and
the default implementations are removed.
BUG=2436
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5222 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-05 12:01:45 +00:00 |
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henrik.lundin@webrtc.org
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9fe3603dc1
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Renaming ViEEncoderObserver::VideoSuspended
New name is ViEEncoderObserver::SuspendChange.
BUG=2436
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5157 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-11-21 23:00:40 +00:00 |
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henrik.lundin@webrtc.org
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ce8e0936d9
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Rename AutoMute to SuspendBelowMinBitrate
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-11-18 12:18:43 +00:00 |
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fischman@webrtc.org
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b7a171825b
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Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5044 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-28 17:36:59 +00:00 |
|
henrik.lundin@webrtc.org
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1a3a6e5340
|
Removing the threshold from the auto-mute APIs
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.
BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-28 10:16:14 +00:00 |
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fischman@webrtc.org
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37bb4974e7
|
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-23 23:59:45 +00:00 |
|
henrik.lundin@webrtc.org
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0d19ed9a06
|
AutoMute: Adding channel_id parameter to callback.
BUG=2436
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2390004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5006 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-21 12:37:13 +00:00 |
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henrik.lundin@webrtc.org
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70df305760
|
Minor fix to avoid breakage
Related to AutoMute feature. Fixed a lint nit, too.
TBR=mflodman@webrtc.org
BUG=2436
Review URL: https://webrtc-codereview.appspot.com/2347004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4910 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-03 13:38:59 +00:00 |
|
henrik.lundin@webrtc.org
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7ea4f24ea5
|
Piping AutoMuter interface through to ViE API
This is a piece of the AutoMuter effort. A second CL will follow containing modifications to the new API, and tests.
BUG=2436
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2331004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4899 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-02 13:34:26 +00:00 |
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elham@webrtc.org
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814e28413d
|
Revert r4562
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2117004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4623 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-08-26 23:21:03 +00:00 |
|
agalusza@google.com
|
b655985abd
|
Added choice of decode error mode to loopback test.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1997004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4562 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-08-16 23:07:14 +00:00 |
|
mflodman@webrtc.org
|
1c986e7c89
|
Removed ViE file API.
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1723004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-06-26 09:12:49 +00:00 |
|
pbos@webrtc.org
|
f5d4cb1958
|
Include files from webrtc/.. paths in video_engine/
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
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2013-05-17 13:44:48 +00:00 |
|
mflodman@webrtc.org
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4aee6b637d
|
Added API to get receive side video delay.
BUG=1222
Review URL: https://webrtc-codereview.appspot.com/997004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3294 4adac7df-926f-26a2-2b94-8c16560cd09d
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2012-12-14 14:02:10 +00:00 |
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andrew@webrtc.org
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14b43beb7c
|
Move src/ -> webrtc/
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
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2012-10-22 18:19:23 +00:00 |
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