Callback for send bitrate estimates - new roll

Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.

Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.

The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:

webrtc::RTPSender::BitrateUpdated()  // Get RTPSender stats lock
webrtc::Bitrate::Process()  // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...

webrtc::Bitrate::Update()  // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats()  // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...

This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.

BUG=2235
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h

@@ -300,6 +300,13 @@ 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.