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Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.

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R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2014-07-08 14:32:58 +00:00
parent 3d7da88e06
commit d11bec40b2
8 changed files with 57 additions and 61 deletions
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8
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
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@ -67,6 +67,7 @@ class RtpRtcp : public Module {
RtpAudioFeedback* audio_messages;
RemoteBitrateEstimator* remote_bitrate_estimator;
PacedSender* paced_sender;
BitrateStatisticsObserver* send_bitrate_observer;
};
/*
@ -308,13 +309,6 @@ class RtpRtcp : public Module {
uint32_t* fecRate,
uint32_t* nackRate) const = 0;
/*
* Called on any new send bitrate estimate.
*/
virtual void RegisterVideoBitrateObserver(
BitrateStatisticsObserver* observer) = 0;
virtual BitrateStatisticsObserver* GetVideoBitrateObserver() const = 0;
/*
* Used by the codec module to deliver a video or audio frame for
* packetization.