We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
