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8f27fcce79
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Revert 8028 "Support associated payload type when registering Rt..."
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-01-09 20:22:46 +00:00 |
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2a169640a3
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Support associated payload type when registering Rtx payload type.
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-01-09 15:16:10 +00:00 |
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931e3da8f2
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Log formatting fix for VideoEncoderConfig.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7648 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-11-06 09:35:08 +00:00 |
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b7ed7799e7
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Implement conference-mode temporal-layer screencast.
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-10-31 13:08:10 +00:00 |
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ad3b5a5c16
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Move min transmit bitrate to VideoEncoderConfig.
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-10-24 09:23:21 +00:00 |
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1e92b0a93d
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Add ToString() to VideoSendStream::Config.
Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.
BUG=3171
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-05-15 09:35:06 +00:00 |
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c49d5b7df8
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Move implementation files out of the webrtc/ root.
Leaves the root for public headers. Also fixes the issue of requiring
root OWNERS approval for changes in the Call implementation and adding
end-to-end tests.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5223 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-05 12:11:47 +00:00 |
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ce90eff345
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Rename RTP-extension constants.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-11-20 11:48:56 +00:00 |
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