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Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.

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R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2014-07-10 09:39:23 +00:00
parent 5bde66e913
commit 8f1512140e
8 changed files with 34 additions and 56 deletions
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15
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -38,7 +38,8 @@ RtpRtcp::Configuration::Configuration()
audio_messages(NullObjectRtpAudioFeedback()),
remote_bitrate_estimator(NULL),
paced_sender(NULL),
send_bitrate_observer(NULL) {
send_bitrate_observer(NULL),
send_frame_count_observer(NULL) {
}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
@ -62,7 +63,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.outgoing_transport,
configuration.audio_messages,
configuration.paced_sender,
configuration.send_bitrate_observer),
configuration.send_bitrate_observer,
configuration.send_frame_count_observer),
rtcp_sender_(configuration.id,
configuration.audio,
configuration.clock,
@ -1350,15 +1352,6 @@ StreamDataCountersCallback*
return rtp_sender_.GetRtpStatisticsCallback();
}
void ModuleRtpRtcpImpl::RegisterSendFrameCountObserver(
FrameCountObserver* observer) {
rtp_sender_.RegisterFrameCountObserver(observer);
}
FrameCountObserver* ModuleRtpRtcpImpl::GetSendFrameCountObserver() const {
return rtp_sender_.GetFrameCountObserver();
}
bool ModuleRtpRtcpImpl::IsDefaultModule() const {
CriticalSectionScoped cs(critical_section_module_ptrs_.get());
return !child_modules_.empty();