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

webrtc/pc/webrtcsession.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 612 matching lines @@
 
   if (options.disable_encryption) {
     webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
   }
 
   return true;
 }
 
 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_);
   RTC_DCHECK(!sctp_transport_);
 }
 
 cricket::BaseChannel* WebRtcSession::GetChannel(
     const std::string& content_name) {
   if (voice_channel() && voice_channel()->content_name() == content_name) {
     return voice_channel();
@@ skipping 1244 matching lines @@
         this, &WebRtcSession::OnDtlsSrtpSetupFailure);
     rtp_data_channel_->SignalSentPacket.connect(this,
                                                 &WebRtcSession::OnSentPacket_w);
   }
 
   SignalDataChannelCreated();
 
   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());
   std::unique_ptr<SessionStats> session_stats(new SessionStats());
   for (const auto channel_name_pair : { &channel_name_pairs.voice,
                                         &channel_name_pairs.video,
                                         &channel_name_pairs.data }) {
     if (*channel_name_pair) {
       cricket::TransportStats transport_stats;
       if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
@@ skipping 402 matching lines @@
                                                   srtp_crypto_suite);
   }
   if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
     metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type,
                                                   ssl_cipher_suite);
   }
 }
 
 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(
     const std::string& content_name) {
   cricket::BaseChannel* channel = GetChannel(content_name);
   if (!channel) {
 #ifdef HAVE_QUIC
     if (data_channel_type_ == cricket::DCT_QUIC && quic_data_transport_ &&
         content_name == quic_data_transport_->transport_name()) {
       return quic_data_transport_->transport_name();
@@ skipping 57 matching lines @@
       (rtp_data_channel_->rtcp_dtls_transport() != nullptr);
   channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
   transport_controller_->DestroyDtlsTransport(
       transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   if (need_to_delete_rtcp) {
     transport_controller_->DestroyDtlsTransport(
         transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 }
 }  // namespace webrtc