Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
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
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@ -37,7 +37,8 @@ RtpRtcp::Configuration::Configuration()
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rtt_stats(NULL),
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audio_messages(NullObjectRtpAudioFeedback()),
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remote_bitrate_estimator(NULL),
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paced_sender(NULL) {
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paced_sender(NULL),
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send_bitrate_observer(NULL) {
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}
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RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
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@ -60,7 +61,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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configuration.clock,
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configuration.outgoing_transport,
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configuration.audio_messages,
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configuration.paced_sender),
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configuration.paced_sender,
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configuration.send_bitrate_observer),
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rtcp_sender_(configuration.id,
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configuration.audio,
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configuration.clock,
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@ -1234,16 +1236,6 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
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*nack_rate = rtp_sender_.NackOverheadRate();
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}
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void ModuleRtpRtcpImpl::RegisterVideoBitrateObserver(
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BitrateStatisticsObserver* observer) {
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assert(!IsDefaultModule());
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rtp_sender_.RegisterBitrateObserver(observer);
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}
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BitrateStatisticsObserver* ModuleRtpRtcpImpl::GetVideoBitrateObserver() const {
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return rtp_sender_.GetBitrateObserver();
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}
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void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
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RequestKeyFrame();
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}
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