diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 7e5cc74..1d9882e 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -171,6 +171,8 @@
 // endpoint that only signals a=msid lines to convey stream_ids.
 void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
 
+// TODO(https://crbug.com/webrtc/14175): Stop depending on "track" stats, the
+// metrics we're interested in are already available in "inbound-rtp".
 int FindFirstMediaStatsIndexByKind(
     const std::string& kind,
     const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
@@ -650,34 +652,26 @@
   void StartWatchingDelayStats() {
     // Get the baseline numbers for audio_packets and audio_delay.
     auto received_stats = NewGetStats();
-    auto track_stats =
-        received_stats->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>()[0];
-    ASSERT_TRUE(track_stats->relative_packet_arrival_delay.is_defined());
     auto rtp_stats =
         received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0];
+    ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
     ASSERT_TRUE(rtp_stats->packets_received.is_defined());
     ASSERT_TRUE(rtp_stats->track_id.is_defined());
-    audio_track_stats_id_ = track_stats->id();
-    ASSERT_TRUE(received_stats->Get(audio_track_stats_id_));
     rtp_stats_id_ = rtp_stats->id();
-    ASSERT_EQ(audio_track_stats_id_, *rtp_stats->track_id);
     audio_packets_stat_ = *rtp_stats->packets_received;
-    audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
-    audio_samples_stat_ = *track_stats->total_samples_received;
-    audio_concealed_stat_ = *track_stats->concealed_samples;
+    audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
+    audio_samples_stat_ = *rtp_stats->total_samples_received;
+    audio_concealed_stat_ = *rtp_stats->concealed_samples;
   }
 
   void UpdateDelayStats(std::string tag, int desc_size) {
     auto report = NewGetStats();
-    auto track_stats =
-        report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id_);
-    ASSERT_TRUE(track_stats);
     auto rtp_stats =
         report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_);
     ASSERT_TRUE(rtp_stats);
     auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
     auto delta_rpad =
-        *track_stats->relative_packet_arrival_delay - audio_delay_stat_;
+        *rtp_stats->relative_packet_arrival_delay - audio_delay_stat_;
     auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
     // The purpose of these checks is to sound the alarm early if we introduce
     // serious regressions. The numbers are not acceptable for production, but
@@ -694,9 +688,9 @@
     EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size;
 #endif
     auto delta_samples =
-        *track_stats->total_samples_received - audio_samples_stat_;
+        *rtp_stats->total_samples_received - audio_samples_stat_;
     auto delta_concealed =
-        *track_stats->concealed_samples - audio_concealed_stat_;
+        *rtp_stats->concealed_samples - audio_concealed_stat_;
     // These limits should be adjusted down as we improve:
     //
     // Concealing more than 4000 samples during a renegotiation is unacceptable.
@@ -731,9 +725,9 @@
     }
     // Increment trailing counters
     audio_packets_stat_ = *rtp_stats->packets_received;
-    audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
-    audio_samples_stat_ = *track_stats->total_samples_received;
-    audio_concealed_stat_ = *track_stats->concealed_samples;
+    audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
+    audio_samples_stat_ = *rtp_stats->total_samples_received;
+    audio_concealed_stat_ = *rtp_stats->concealed_samples;
   }
 
   // Sets number of candidates expected
@@ -1240,7 +1234,6 @@
   uint64_t audio_samples_stat_ = 0;
   uint64_t audio_concealed_stat_ = 0;
   std::string rtp_stats_id_;
-  std::string audio_track_stats_id_;
 
   ScopedTaskSafety task_safety_;