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

webrtc/pc/rtcstatscollector_unittest.cc

Index: webrtc/pc/rtcstatscollector_unittest.cc
diff --git a/webrtc/pc/rtcstatscollector_unittest.cc b/webrtc/pc/rtcstatscollector_unittest.cc
index 1940da69daed9e4f519ea39e287f0360b3184a50..26d0339e4f93344e124f94e52fedffa8b9ba0708 100644
--- a/webrtc/pc/rtcstatscollector_unittest.cc
+++ b/webrtc/pc/rtcstatscollector_unittest.cc
@@ -581,7 +581,9 @@ class FakeRTCStatsCollector : public RTCStatsCollector,
         new RTCTestStats("SignalingThreadStats", timestamp_us)));
     AddPartialResults(signaling_report);
   }
-  void ProducePartialResultsOnNetworkThread(int64_t timestamp_us) override {
+  void ProducePartialResultsOnNetworkThread(
+      int64_t timestamp_us,
+      const Call::Stats& call_stats) override {
     EXPECT_TRUE(network_thread_->IsCurrent());
     {
       rtc::CritScope cs(&lock_);
@@ -1263,9 +1265,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
   // Mock the session to return bandwidth estimation info. These should only
   // be used for a selected candidate pair.
   cricket::VideoMediaInfo video_media_info;
-  video_media_info.bw_estimations.push_back(cricket::BandwidthEstimationInfo());
-  video_media_info.bw_estimations[0].available_send_bandwidth = 8888;
-  video_media_info.bw_estimations[0].available_recv_bandwidth = 9999;
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
   EXPECT_CALL(test_->session(), video_channel())
@@ -1345,8 +1344,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
       .channel_stats[0]
       .connection_infos[0]
       .best_connection = true;
-  video_media_info.bw_estimations[0].available_send_bandwidth = 0;
-  video_media_info.bw_estimations[0].available_recv_bandwidth = 0;
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
   collector_->ClearCachedStatsReport();
@@ -1360,14 +1357,19 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
   EXPECT_TRUE(report->Get(*expected_pair.transport_id));
 
   // Set bandwidth and "GetStats" again.
-  video_media_info.bw_estimations[0].available_send_bandwidth = 888;
-  video_media_info.bw_estimations[0].available_recv_bandwidth = 999;
+  webrtc::Call::Stats call_stats;
+  const int kSendBandwidth = 888;
+  call_stats.send_bandwidth_bps = kSendBandwidth;
+  const int kRecvBandwidth = 999;
+  call_stats.recv_bandwidth_bps = kRecvBandwidth;
+  EXPECT_CALL(test_->session(), GetCallStats())
+      .WillRepeatedly(Return(call_stats));
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
   collector_->ClearCachedStatsReport();
   report = GetStatsReport();
-  expected_pair.available_outgoing_bitrate = 888;
-  expected_pair.available_incoming_bitrate = 999;
+  expected_pair.available_outgoing_bitrate = kSendBandwidth;
+  expected_pair.available_incoming_bitrate = kRecvBandwidth;
   ASSERT_TRUE(report->Get(expected_pair.id()));
   EXPECT_EQ(
       expected_pair,