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

webrtc/pc/rtcstatscollector.h

 /*
  *  Copyright 2016 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.
  */
 
 #ifndef WEBRTC_PC_RTCSTATSCOLLECTOR_H_
 #define WEBRTC_PC_RTCSTATSCOLLECTOR_H_
 
 #include <map>
 #include <memory>
 #include <set>
 #include <vector>
 
 #include "webrtc/api/stats/rtcstats_objects.h"
 #include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/stats/rtcstatsreport.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/timeutils.h"
+#include "webrtc/call/call.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/pc/datachannel.h"
 #include "webrtc/pc/trackmediainfomap.h"
 
 namespace cricket {
 class Candidate;
 }  // namespace cricket
 
 namespace rtc {
 class SSLCertificate;
@@ skipping 58 matching lines @@
       RTCStatsReport* report) const;
   // Produces |RTCCodecStats|.
   void ProduceCodecStats_n(
       int64_t timestamp_us, const TrackMediaInfoMap& track_media_info_map,
       RTCStatsReport* report) const;
   // Produces |RTCDataChannelStats|.
   void ProduceDataChannelStats_s(
       int64_t timestamp_us, RTCStatsReport* report) const;
   // Produces |RTCIceCandidatePairStats| and |RTCIceCandidateStats|.
   void ProduceIceCandidateAndPairStats_n(
-      int64_t timestamp_us, const SessionStats& session_stats,
+      int64_t timestamp_us,
+      const SessionStats& session_stats,
       const cricket::VideoMediaInfo* video_media_info,
+      const Call::Stats& call_stats,
       RTCStatsReport* report) const;
   // Produces |RTCMediaStreamStats| and |RTCMediaStreamTrackStats|.
   void ProduceMediaStreamAndTrackStats_s(
       int64_t timestamp_us, RTCStatsReport* report) const;
   // Produces |RTCPeerConnectionStats|.
   void ProducePeerConnectionStats_s(
       int64_t timestamp_us, RTCStatsReport* report) const;
   // Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
   void ProduceRTPStreamStats_n(
       int64_t timestamp_us, const SessionStats& session_stats,
@@ skipping 28 matching lines @@
   rtc::scoped_refptr<RTCStatsReport> partial_report_;
   std::vector<rtc::scoped_refptr<RTCStatsCollectorCallback>> callbacks_;
 
   // Set in |GetStatsReport|, read in |ProducePartialResultsOnNetworkThread| and
   // |ProducePartialResultsOnSignalingThread|, reset after work is complete. Not
   // passed as arguments to avoid copies. This is thread safe - when we
   // set/reset we know there are no pending stats requests in progress.
   std::unique_ptr<ChannelNamePairs> channel_name_pairs_;
   std::unique_ptr<TrackMediaInfoMap> track_media_info_map_;
   std::map<MediaStreamTrackInterface*, std::string> track_to_id_;
+  Call::Stats call_stats_;
 
   // A timestamp, in microseconds, that is based on a timer that is
   // monotonically increasing. That is, even if the system clock is modified the
   // difference between the timer and this timestamp is how fresh the cached
   // report is.
   int64_t cache_timestamp_us_;
   int64_t cache_lifetime_us_;
   rtc::scoped_refptr<const RTCStatsReport> cached_report_;
 
   // Data recorded and maintained by the stats collector during its lifetime.
@@ skipping 17 matching lines @@
 };
 
 const char* CandidateTypeToRTCIceCandidateTypeForTesting(
     const std::string& type);
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTCSTATSCOLLECTOR_H_