Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.

webrtc/media/engine/webrtcvideoengine2.h

Index: webrtc/media/engine/webrtcvideoengine2.h
diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
index 394646bc3278bee5d1e52fcd77b067d958281953..56431167a8397f06b4f1c70de3df8cfd06d18231 100644
--- a/webrtc/media/engine/webrtcvideoengine2.h
+++ b/webrtc/media/engine/webrtcvideoengine2.h
@@ -160,6 +160,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
   bool RemoveRecvStream(uint32_t ssrc) override;
   bool SetSink(uint32_t ssrc,
                rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
   bool GetStats(VideoMediaInfo* info) override;
 
   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
@@ -284,7 +285,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
 
     const std::vector<uint32_t>& GetSsrcs() const;
     VideoSenderInfo GetVideoSenderInfo(bool log_stats);
-    void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
+    void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
 
    private:
     // Parameters needed to reconstruct the underlying stream.