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

webrtc/pc/webrtcsession.cc

Index: webrtc/pc/webrtcsession.cc
diff --git a/webrtc/pc/webrtcsession.cc b/webrtc/pc/webrtcsession.cc
index cc424b222e081c994b6a7e546f59c126d86a7c01..ddf5ba760a7a41c9ec8aaceac91a1b1fbba6f8a8 100644
--- a/webrtc/pc/webrtcsession.cc
+++ b/webrtc/pc/webrtcsession.cc
@@ -630,7 +630,9 @@ bool WebRtcSession::Initialize(
 
 void WebRtcSession::Close() {
   SetState(STATE_CLOSED);
+  call_ = nullptr;
   RemoveUnusedChannels(nullptr);
+  RTC_DCHECK(!call_);
   RTC_DCHECK(!voice_channel_);
   RTC_DCHECK(!video_channel_);
   RTC_DCHECK(!rtp_data_channel_);
@@ -1895,6 +1897,15 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
   return true;
 }
 
+Call::Stats WebRtcSession::GetCallStats() {
+  if (!worker_thread()->IsCurrent()) {
+    return worker_thread()->Invoke<Call::Stats>(
+        RTC_FROM_HERE, rtc::Bind(&WebRtcSession::GetCallStats, this));
+  }
+  RTC_DCHECK(call_);
+  return call_->GetStats();
+}
+
 std::unique_ptr<SessionStats> WebRtcSession::GetStats_n(
     const ChannelNamePairs& channel_name_pairs) {
   RTC_DCHECK(network_thread()->IsCurrent());
@@ -2317,7 +2328,8 @@ void WebRtcSession::ReportNegotiatedCiphers(
 
 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
-  call_->OnSentPacket(sent_packet);
+  if (call_)
+    call_->OnSentPacket(sent_packet);
 }
 
 const std::string WebRtcSession::GetTransportName(