1 files changed, 22 insertions, 14 deletions

diff --git a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
index 055be6fe99..61b3bca1db 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
@@ -29,6 +29,7 @@
 #include "api/media_stream_track.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
+#include "api/stats/attribute.h"
 #include "api/stats/rtc_stats.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/stats/rtcstats_objects.h"
@@ -2303,7 +2304,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) {
     ASSERT_TRUE(report->Get("ITTransportName1A1"));
     auto stats =
         report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
-    ASSERT_FALSE(stats.playout_id.is_defined());
+    ASSERT_FALSE(stats.playout_id.has_value());
   }
   {
     // We do expect a playout id when receiving.
@@ -2314,7 +2315,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) {
     ASSERT_TRUE(report->Get("ITTransportName1A1"));
     auto stats =
         report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
-    ASSERT_TRUE(stats.playout_id.is_defined());
+    ASSERT_TRUE(stats.playout_id.has_value());
     EXPECT_EQ(*stats.playout_id, "AP");
   }
 }
@@ -2478,6 +2479,10 @@ TEST_F(RTCStatsCollectorTest, CollectRTCAudioPlayoutStats) {
   audio_device_stats.total_playout_delay_s = 5;
   pc_->SetAudioDeviceStats(audio_device_stats);
 
+  pc_->AddVoiceChannel("AudioMid", "TransportName", {});
+  stats_->SetupRemoteTrackAndReceiver(
+      cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
+
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
   auto stats_of_track_type = report->GetStatsOfType<RTCAudioPlayoutStats>();
   ASSERT_EQ(1U, stats_of_track_type.size());
@@ -2526,7 +2531,7 @@ TEST_F(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
   auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   ASSERT_EQ(inbound_rtps.size(), 1u);
-  ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.is_defined());
+  ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.has_value());
   EXPECT_EQ(*inbound_rtps[0]->goog_timing_frame_info,
             "1,2,3,4,5,6,7,8,9,10,11,12,13,1,0");
 }
@@ -3135,8 +3140,8 @@ TEST_F(RTCStatsCollectorTest,
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
   ASSERT_TRUE(report->Get("SV42"));
   auto video_stats = report->Get("SV42")->cast_to<RTCVideoSourceStats>();
-  EXPECT_FALSE(video_stats.frames_per_second.is_defined());
-  EXPECT_FALSE(video_stats.frames.is_defined());
+  EXPECT_FALSE(video_stats.frames_per_second.has_value());
+  EXPECT_FALSE(video_stats.frames.has_value());
 }
 
 // The track not having a source is not expected to be true in practise, but
@@ -3165,8 +3170,8 @@ TEST_F(RTCStatsCollectorTest,
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
   ASSERT_TRUE(report->Get("SV42"));
   auto video_stats = report->Get("SV42")->cast_to<RTCVideoSourceStats>();
-  EXPECT_FALSE(video_stats.width.is_defined());
-  EXPECT_FALSE(video_stats.height.is_defined());
+  EXPECT_FALSE(video_stats.width.has_value());
+  EXPECT_FALSE(video_stats.height.has_value());
 }
 
 TEST_F(RTCStatsCollectorTest,
@@ -3367,9 +3372,9 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
   auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                  ->cast_to<RTCRemoteInboundRtpStreamStats>();
 
-  EXPECT_TRUE(remote_inbound_rtp.round_trip_time_measurements.is_defined());
+  EXPECT_TRUE(remote_inbound_rtp.round_trip_time_measurements.has_value());
   EXPECT_EQ(0, *remote_inbound_rtp.round_trip_time_measurements);
-  EXPECT_FALSE(remote_inbound_rtp.round_trip_time.is_defined());
+  EXPECT_FALSE(remote_inbound_rtp.round_trip_time.has_value());
 }
 
 TEST_P(RTCStatsCollectorTestWithParamKind,
@@ -3431,10 +3436,10 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
   auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                  ->cast_to<RTCRemoteInboundRtpStreamStats>();
 
-  EXPECT_TRUE(remote_inbound_rtp.codec_id.is_defined());
+  EXPECT_TRUE(remote_inbound_rtp.codec_id.has_value());
   EXPECT_TRUE(report->Get(*remote_inbound_rtp.codec_id));
 
-  EXPECT_TRUE(remote_inbound_rtp.jitter.is_defined());
+  EXPECT_TRUE(remote_inbound_rtp.jitter.has_value());
   // The jitter (in seconds) is the report block's jitter divided by the codec's
   // clock rate.
   EXPECT_EQ(5.0, *remote_inbound_rtp.jitter);
@@ -3471,7 +3476,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
   auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                  ->cast_to<RTCRemoteInboundRtpStreamStats>();
 
-  EXPECT_TRUE(remote_inbound_rtp.transport_id.is_defined());
+  EXPECT_TRUE(remote_inbound_rtp.transport_id.has_value());
   EXPECT_EQ("TTransportName2",  // 2 for RTCP
             *remote_inbound_rtp.transport_id);
   EXPECT_TRUE(report->Get(*remote_inbound_rtp.transport_id));
@@ -3716,12 +3721,15 @@ class RTCTestStats : public RTCStats {
   WEBRTC_RTCSTATS_DECL();
 
   RTCTestStats(const std::string& id, Timestamp timestamp)
-      : RTCStats(id, timestamp), dummy_stat("dummyStat") {}
+      : RTCStats(id, timestamp) {}
 
   RTCStatsMember<int32_t> dummy_stat;
 };
 
-WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats", &dummy_stat)
+WEBRTC_RTCSTATS_IMPL(RTCTestStats,
+                     RTCStats,
+                     "test-stats",
+                     AttributeInit("dummyStat", &dummy_stat))
 
 // Overrides the stats collection to verify thread usage and that the resulting
 // partial reports are merged.