1 files changed, 12 insertions, 12 deletions

diff --git a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
index bfff86ee93..c960a36b5e 100644
--- a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
@@ -1356,15 +1356,15 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
   ASSERT_EQ(outbound_stream_stats.size(), 4u);
   std::vector<std::string> outbound_track_ids;
   for (const auto& stat : outbound_stream_stats) {
-    ASSERT_TRUE(stat->bytes_sent.is_defined());
+    ASSERT_TRUE(stat->bytes_sent.has_value());
     EXPECT_LT(0u, *stat->bytes_sent);
     if (*stat->kind == "video") {
-      ASSERT_TRUE(stat->key_frames_encoded.is_defined());
+      ASSERT_TRUE(stat->key_frames_encoded.has_value());
       EXPECT_GT(*stat->key_frames_encoded, 0u);
-      ASSERT_TRUE(stat->frames_encoded.is_defined());
+      ASSERT_TRUE(stat->frames_encoded.has_value());
       EXPECT_GE(*stat->frames_encoded, *stat->key_frames_encoded);
     }
-    ASSERT_TRUE(stat->media_source_id.is_defined());
+    ASSERT_TRUE(stat->media_source_id.has_value());
     const RTCMediaSourceStats* media_source =
         static_cast<const RTCMediaSourceStats*>(
             caller_report->Get(*stat->media_source_id));
@@ -1381,12 +1381,12 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
   ASSERT_EQ(4u, inbound_stream_stats.size());
   std::vector<std::string> inbound_track_ids;
   for (const auto& stat : inbound_stream_stats) {
-    ASSERT_TRUE(stat->bytes_received.is_defined());
+    ASSERT_TRUE(stat->bytes_received.has_value());
     EXPECT_LT(0u, *stat->bytes_received);
     if (*stat->kind == "video") {
-      ASSERT_TRUE(stat->key_frames_decoded.is_defined());
+      ASSERT_TRUE(stat->key_frames_decoded.has_value());
       EXPECT_GT(*stat->key_frames_decoded, 0u);
-      ASSERT_TRUE(stat->frames_decoded.is_defined());
+      ASSERT_TRUE(stat->frames_decoded.has_value());
       EXPECT_GE(*stat->frames_decoded, *stat->key_frames_decoded);
     }
     inbound_track_ids.push_back(*stat->track_identifier);
@@ -1417,7 +1417,7 @@ TEST_P(PeerConnectionIntegrationTest,
   auto inbound_stream_stats =
       report->GetStatsOfType<RTCInboundRtpStreamStats>();
   ASSERT_EQ(1U, inbound_stream_stats.size());
-  ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
+  ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.has_value());
   ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
 }
 
@@ -1464,7 +1464,7 @@ TEST_P(PeerConnectionIntegrationTest,
   auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
   ASSERT_GE(index, 0);
-  EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
+  EXPECT_TRUE(inbound_rtps[index]->audio_level.has_value());
 }
 
 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
@@ -2952,7 +2952,7 @@ double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
   auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   RTC_CHECK(!inbound_rtps.empty());
   auto* inbound_rtp = inbound_rtps[0];
-  if (!inbound_rtp->total_audio_energy.is_defined()) {
+  if (!inbound_rtp->total_audio_energy.has_value()) {
     return 0.0;
   }
   return *inbound_rtp->total_audio_energy;
@@ -3776,7 +3776,7 @@ int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
     ADD_FAILURE();
     return 0;
   }
-  if (!sender_stats[0]->nack_count.is_defined()) {
+  if (!sender_stats[0]->nack_count.has_value()) {
     return 0;
   }
   return *sender_stats[0]->nack_count;
@@ -3789,7 +3789,7 @@ int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
     ADD_FAILURE();
     return 0;
   }
-  if (!receiver_stats[0]->nack_count.is_defined()) {
+  if (!receiver_stats[0]->nack_count.has_value()) {
     return 0;
   }
   return *receiver_stats[0]->nack_count;