diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
index 5d7f8aa2c36..7a82666e812 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
@@ -124,7 +124,7 @@ void LossBasedBandwidthEstimation::UpdateLossStatistics(
     return;
   }
   int loss_count = 0;
-  for (auto pkt : packet_results) {
+  for (const auto& pkt : packet_results) {
     loss_count += pkt.receive_time.IsInfinite() ? 1 : 0;
   }
   last_loss_ratio_ = static_cast<double>(loss_count) / packet_results.size();
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 383f785dfea..08405046528 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -437,7 +437,7 @@ void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
     packet_information->packet_type_flags |= kRtcpRr;
   }
 
-  for (const rtcp::ReportBlock report_block : sender_report.report_blocks())
+  for (const rtcp::ReportBlock& report_block : sender_report.report_blocks())
     HandleReportBlock(report_block, packet_information, remote_ssrc);
 }
 
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
index f14a0d117e3..321d21d35bd 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
@@ -353,7 +353,7 @@ bool MdnsMessage::Write(rtc::ByteBufferWriter* buf) const {
   header_.Write(buf);
 
   auto write_rr = [&buf](const std::vector<MdnsResourceRecord>& section) {
-    for (auto rr : section) {
+    for (const auto& rr : section) {
       if (!rr.Write(buf)) {
         return false;
       }
@@ -361,7 +361,7 @@ bool MdnsMessage::Write(rtc::ByteBufferWriter* buf) const {
     return true;
   };
 
-  for (auto question : question_section_) {
+  for (const auto& question : question_section_) {
     if (!question.Write(buf)) {
       return false;
     }
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
index bb20a4d5da3..a180180b91d 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
@@ -948,7 +948,7 @@ void Port::UpdateNetworkCost() {
   // Network cost change will affect the connection selection criteria.
   // Signal the connection state change on each connection to force a
   // re-sort in P2PTransportChannel.
-  for (auto kv : connections_) {
+  for (const auto& kv : connections_) {
     Connection* conn = kv.second;
     conn->SignalStateChange(conn);
   }
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
index edba4d68e54..dfb27fccb17 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
@@ -69,7 +69,7 @@ void StunRequestManager::SendDelayed(StunRequest* request, int delay) {
 }
 
 void StunRequestManager::Flush(int msg_type) {
-  for (const auto kv : requests_) {
+  for (const auto& kv : requests_) {
     StunRequest* request = kv.second;
     if (msg_type == kAllRequests || msg_type == request->type()) {
       thread_->Clear(request, MSG_STUN_SEND);
@@ -79,7 +79,7 @@ void StunRequestManager::Flush(int msg_type) {
 }
 
 bool StunRequestManager::HasRequest(int msg_type) {
-  for (const auto kv : requests_) {
+  for (const auto& kv : requests_) {
     StunRequest* request = kv.second;
     if (msg_type == kAllRequests || msg_type == request->type()) {
       return true;
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
index 78ecaf31dea..328e3ca42b1 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
@@ -629,7 +629,7 @@ RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroup(
 
   // The BUNDLE group containing a MID that no m= section has is invalid.
   if (new_bundle_group) {
-    for (auto content_name : new_bundle_group->content_names()) {
+    for (const auto& content_name : new_bundle_group->content_names()) {
       if (!description->GetContentByName(content_name)) {
         return RTCError(RTCErrorType::INVALID_PARAMETER,
                         "The BUNDLE group contains MID:" + content_name +
@@ -645,7 +645,7 @@ RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroup(
 
     if (new_bundle_group) {
       // The BUNDLE group in answer should be a subset of offered group.
-      for (auto content_name : new_bundle_group->content_names()) {
+      for (const auto& content_name : new_bundle_group->content_names()) {
         if (!offered_bundle_group ||
             !offered_bundle_group->HasContentName(content_name)) {
           return RTCError(RTCErrorType::INVALID_PARAMETER,
@@ -656,7 +656,7 @@ RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroup(
     }
 
     if (bundle_group_) {
-      for (auto content_name : bundle_group_->content_names()) {
+      for (const auto& content_name : bundle_group_->content_names()) {
         // An answer that removes m= sections from pre-negotiated BUNDLE group
         // without rejecting it, is invalid.
         if (!new_bundle_group ||
@@ -704,7 +704,7 @@ RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroup(
   // If the |bundled_content| is rejected, other contents in the bundle group
   // should be rejected.
   if (bundled_content->rejected) {
-    for (auto content_name : bundle_group_->content_names()) {
+    for (const auto& content_name : bundle_group_->content_names()) {
       auto other_content = description->GetContentByName(content_name);
       if (!other_content->rejected) {
         return RTCError(
@@ -739,7 +739,7 @@ void JsepTransportController::HandleRejectedContent(
   // then destroy the cricket::JsepTransport.
   RemoveTransportForMid(content_info.name);
   if (content_info.name == bundled_mid()) {
-    for (auto content_name : bundle_group_->content_names()) {
+    for (const auto& content_name : bundle_group_->content_names()) {
       RemoveTransportForMid(content_name);
     }
     bundle_group_.reset();
@@ -845,7 +845,7 @@ std::vector<int> JsepTransportController::GetEncryptedHeaderExtensionIds(
   }
 
   std::vector<int> encrypted_header_extension_ids;
-  for (auto extension : content_desc->rtp_header_extensions()) {
+  for (const auto& extension : content_desc->rtp_header_extensions()) {
     if (!extension.encrypt) {
       continue;
     }
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
index 4f0a8ea0112..ee8a909e03e 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
@@ -832,7 +832,7 @@ PeerConnection::~PeerConnection() {
   // Need to stop transceivers before destroying the stats collector because
   // AudioRtpSender has a reference to the StatsCollector it will update when
   // stopping.
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     transceiver->Stop();
   }
 
@@ -868,12 +868,12 @@ PeerConnection::~PeerConnection() {
 void PeerConnection::DestroyAllChannels() {
   // Destroy video channels first since they may have a pointer to a voice
   // channel.
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
       DestroyTransceiverChannel(transceiver);
     }
   }
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
       DestroyTransceiverChannel(transceiver);
     }
@@ -1611,7 +1611,7 @@ rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
     const {
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
-  for (auto sender : GetSendersInternal()) {
+  for (const auto& sender : GetSendersInternal()) {
     ret.push_back(sender);
   }
   return ret;
@@ -1621,7 +1621,7 @@ std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
 PeerConnection::GetSendersInternal() const {
   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
       all_senders;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     auto senders = transceiver->internal()->senders();
     all_senders.insert(all_senders.end(), senders.begin(), senders.end());
   }
@@ -1643,7 +1643,7 @@ PeerConnection::GetReceiversInternal() const {
   std::vector<
       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
       all_receivers;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     auto receivers = transceiver->internal()->receivers();
     all_receivers.insert(all_receivers.end(), receivers.begin(),
                          receivers.end());
@@ -1656,7 +1656,7 @@ PeerConnection::GetTransceivers() const {
   RTC_CHECK(IsUnifiedPlan())
       << "GetTransceivers is only supported with Unified Plan SdpSemantics.";
   std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     all_transceivers.push_back(transceiver);
   }
   return all_transceivers;
@@ -1867,7 +1867,7 @@ RTCError PeerConnection::HandleLegacyOfferOptions(
 
 void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
     cricket::MediaType media_type) {
-  for (auto transceiver : GetReceivingTransceiversOfType(media_type)) {
+  for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
     RtpTransceiverDirection new_direction =
         RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
     if (new_direction != transceiver->direction()) {
@@ -1901,7 +1901,7 @@ PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
   std::vector<
       rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
       receiving_transceivers;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     if (!transceiver->stopped() && transceiver->media_type() == media_type &&
         RtpTransceiverDirectionHasRecv(transceiver->direction())) {
       receiving_transceivers.push_back(transceiver);
@@ -2094,7 +2094,7 @@ RTCError PeerConnection::ApplyLocalDescription(
     }
     std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
     std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
-    for (auto transceiver : transceivers_) {
+    for (const auto& transceiver : transceivers_) {
       const ContentInfo* content =
           FindMediaSectionForTransceiver(transceiver, local_description());
       if (!content) {
@@ -2122,10 +2122,10 @@ RTCError PeerConnection::ApplyLocalDescription(
       }
     }
     auto observer = Observer();
-    for (auto transceiver : remove_list) {
+    for (const auto& transceiver : remove_list) {
       observer->OnRemoveTrack(transceiver->receiver());
     }
-    for (auto stream : removed_streams) {
+    for (const auto& stream : removed_streams) {
       observer->OnRemoveStream(stream);
     }
   } else {
@@ -2167,7 +2167,7 @@ RTCError PeerConnection::ApplyLocalDescription(
   }
 
   if (IsUnifiedPlan()) {
-    for (auto transceiver : transceivers_) {
+    for (const auto& transceiver : transceivers_) {
       const ContentInfo* content =
           FindMediaSectionForTransceiver(transceiver, local_description());
       if (!content) {
@@ -2447,7 +2447,7 @@ RTCError PeerConnection::ApplyRemoteDescription(
     std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
     std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
     std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
-    for (auto transceiver : transceivers_) {
+    for (const auto& transceiver : transceivers_) {
       const ContentInfo* content =
           FindMediaSectionForTransceiver(transceiver, remote_description());
       if (!content) {
@@ -2541,19 +2541,19 @@ RTCError PeerConnection::ApplyRemoteDescription(
     }
     // Once all processing has finished, fire off callbacks.
     auto observer = Observer();
-    for (auto transceiver : now_receiving_transceivers) {
+    for (const auto& transceiver : now_receiving_transceivers) {
       stats_->AddTrack(transceiver->receiver()->track());
       observer->OnTrack(transceiver);
       observer->OnAddTrack(transceiver->receiver(),
                            transceiver->receiver()->streams());
     }
-    for (auto stream : added_streams) {
+    for (const auto& stream : added_streams) {
       observer->OnAddStream(stream);
     }
-    for (auto transceiver : remove_list) {
+    for (const auto& transceiver : remove_list) {
       observer->OnRemoveTrack(transceiver->receiver());
     }
-    for (auto stream : removed_streams) {
+    for (const auto& stream : removed_streams) {
       observer->OnRemoveStream(stream);
     }
   }
@@ -2660,7 +2660,7 @@ void PeerConnection::ProcessRemovalOfRemoteTrack(
   // TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead
   // of streams, see if the stream was removed by checking if this was the
   // last receiver with that stream ID.
-  for (auto stream : media_streams) {
+  for (const auto& stream : media_streams) {
     if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
       remote_streams_->RemoveStream(stream);
       removed_streams->push_back(stream);
@@ -3382,7 +3382,7 @@ void PeerConnection::Close() {
   ChangeSignalingState(PeerConnectionInterface::kClosed);
   NoteUsageEvent(UsageEvent::CLOSE_CALLED);
 
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     transceiver->Stop();
   }
 
@@ -4045,7 +4045,7 @@ void PeerConnection::GetOptionsForUnifiedPlanOffer(
   // and not associated). Reuse media sections marked as recyclable first,
   // otherwise append to the end of the offer. New media sections should be
   // added in the order they were added to the PeerConnection.
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     if (transceiver->mid() || transceiver->stopped()) {
       continue;
     }
@@ -4815,7 +4815,7 @@ bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
 
 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
 PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     for (auto sender : transceiver->internal()->senders()) {
       if (sender->track() == track) {
         return sender;
@@ -4827,7 +4827,7 @@ PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
 
 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
 PeerConnection::FindSenderById(const std::string& sender_id) const {
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     for (auto sender : transceiver->internal()->senders()) {
       if (sender->id() == sender_id) {
         return sender;
@@ -4839,7 +4839,7 @@ PeerConnection::FindSenderById(const std::string& sender_id) const {
 
 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
 PeerConnection::FindReceiverById(const std::string& receiver_id) const {
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     for (auto receiver : transceiver->internal()->receivers()) {
       if (receiver->id() == receiver_id) {
         return receiver;
@@ -4999,7 +4999,7 @@ void PeerConnection::StopRtcEventLog_w() {
 
 cricket::ChannelInterface* PeerConnection::GetChannel(
     const std::string& content_name) {
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     cricket::ChannelInterface* channel = transceiver->internal()->channel();
     if (channel && channel->content_name() == content_name) {
       return channel;
@@ -5114,7 +5114,7 @@ RTCError PeerConnection::PushdownMediaDescription(
   RTC_DCHECK(sdesc);
 
   // Push down the new SDP media section for each audio/video transceiver.
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     const ContentInfo* content_info =
         FindMediaSectionForTransceiver(transceiver, sdesc);
     cricket::ChannelInterface* channel = transceiver->internal()->channel();
@@ -5463,7 +5463,7 @@ cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
 std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
     const {
   std::map<std::string, std::string> transport_names_by_mid;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     cricket::ChannelInterface* channel = transceiver->internal()->channel();
     if (channel) {
       transport_names_by_mid[channel->content_name()] =
@@ -5640,7 +5640,7 @@ void PeerConnection::OnTransportControllerDtlsHandshakeError(
 }
 
 void PeerConnection::EnableSending() {
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     cricket::ChannelInterface* channel = transceiver->internal()->channel();
     if (channel && !channel->enabled()) {
       channel->Enable(true);
@@ -6377,7 +6377,7 @@ void PeerConnection::OnTransportControllerGatheringState(
 void PeerConnection::ReportTransportStats() {
   std::map<std::string, std::set<cricket::MediaType>>
       media_types_by_transport_name;
-  for (auto transceiver : transceivers_) {
+  for (const auto& transceiver : transceivers_) {
     if (transceiver->internal()->channel()) {
       const std::string& transport_name =
           transceiver->internal()->channel()->transport_name();
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
index d48ecc01f35..6fc4118d979 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
@@ -534,7 +534,7 @@ void ProduceSenderMediaTrackStats(
   // TODO(hbos): Return stats of detached tracks. We have to perform stats
   // gathering at the time of detachment to get accurate stats and timestamps.
   // https://crbug.com/659137
-  for (auto sender : senders) {
+  for (const auto& sender : senders) {
     if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
       AudioTrackInterface* track =
           static_cast<AudioTrackInterface*>(sender->track().get());
@@ -602,7 +602,7 @@ void ProduceReceiverMediaTrackStats(
     std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers,
     RTCStatsReport* report) {
   // This function iterates over the receivers to find the remote tracks.
-  for (auto receiver : receivers) {
+  for (const auto& receiver : receivers) {
     if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
       AudioTrackInterface* track =
           static_cast<AudioTrackInterface*>(receiver->track().get());
@@ -1116,7 +1116,7 @@ void RTCStatsCollector::ProduceMediaStreamStats_s(
   std::map<std::string, std::vector<std::string>> track_ids;
 
   for (const auto& stats : transceiver_stats_infos_) {
-    for (auto sender : stats.transceiver->senders()) {
+    for (const auto& sender : stats.transceiver->senders()) {
       std::string track_id =
           RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
               kSender, sender->internal()->AttachmentId());
@@ -1124,7 +1124,7 @@ void RTCStatsCollector::ProduceMediaStreamStats_s(
         track_ids[stream_id].push_back(track_id);
       }
     }
-    for (auto receiver : stats.transceiver->receivers()) {
+    for (const auto& receiver : stats.transceiver->receivers()) {
       std::string track_id =
           RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
               kReceiver, receiver->internal()->AttachmentId());
@@ -1150,14 +1150,14 @@ void RTCStatsCollector::ProduceMediaStreamTrackStats_s(
   RTC_DCHECK(signaling_thread_->IsCurrent());
   for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos_) {
     std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
-    for (auto sender : stats.transceiver->senders()) {
+    for (const auto& sender : stats.transceiver->senders()) {
       senders.push_back(sender->internal());
     }
     ProduceSenderMediaTrackStats(timestamp_us, *stats.track_media_info_map,
                                  senders, report);
 
     std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
-    for (auto receiver : stats.transceiver->receivers()) {
+    for (const auto& receiver : stats.transceiver->receivers()) {
       receivers.push_back(receiver->internal());
     }
     ProduceReceiverMediaTrackStats(timestamp_us, *stats.track_media_info_map,
@@ -1420,7 +1420,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfos_s() const {
            std::unique_ptr<cricket::VideoMediaInfo>>
       video_stats;
 
-  for (auto transceiver : pc_->GetTransceiversInternal()) {
+  for (const auto& transceiver : pc_->GetTransceiversInternal()) {
     cricket::MediaType media_type = transceiver->media_type();
 
     // Prepare stats entry. The TrackMediaInfoMap will be filled in after the
@@ -1493,11 +1493,11 @@ RTCStatsCollector::PrepareTransceiverStatsInfos_s() const {
       }
     }
     std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
-    for (auto sender : transceiver->senders()) {
+    for (const auto& sender : transceiver->senders()) {
       senders.push_back(sender->internal());
     }
     std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
-    for (auto receiver : transceiver->receivers()) {
+    for (const auto& receiver : transceiver->receivers()) {
       receivers.push_back(receiver->internal());
     }
     stats.track_media_info_map = absl::make_unique<TrackMediaInfoMap>(
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
index 1916a736703..e5a4373294c 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
@@ -207,9 +207,9 @@ void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
 void AudioRtpReceiver::SetStreams(
     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
   // Remove remote track from any streams that are going away.
-  for (auto existing_stream : streams_) {
+  for (const auto& existing_stream : streams_) {
     bool removed = true;
-    for (auto stream : streams) {
+    for (const auto& stream : streams) {
       if (existing_stream->id() == stream->id()) {
         RTC_DCHECK_EQ(existing_stream.get(), stream.get());
         removed = false;
@@ -221,9 +221,9 @@ void AudioRtpReceiver::SetStreams(
     }
   }
   // Add remote track to any streams that are new.
-  for (auto stream : streams) {
+  for (const auto& stream : streams) {
     bool added = true;
-    for (auto existing_stream : streams_) {
+    for (const auto& existing_stream : streams_) {
       if (stream->id() == existing_stream->id()) {
         RTC_DCHECK_EQ(stream.get(), existing_stream.get());
         added = false;
@@ -406,9 +406,9 @@ void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
 void VideoRtpReceiver::SetStreams(
     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
   // Remove remote track from any streams that are going away.
-  for (auto existing_stream : streams_) {
+  for (const auto& existing_stream : streams_) {
     bool removed = true;
-    for (auto stream : streams) {
+    for (const auto& stream : streams) {
       if (existing_stream->id() == stream->id()) {
         RTC_DCHECK_EQ(existing_stream.get(), stream.get());
         removed = false;
@@ -420,9 +420,9 @@ void VideoRtpReceiver::SetStreams(
     }
   }
   // Add remote track to any streams that are new.
-  for (auto stream : streams) {
+  for (const auto& stream : streams) {
     bool added = true;
-    for (auto existing_stream : streams_) {
+    for (const auto& existing_stream : streams_) {
       if (stream->id() == existing_stream->id()) {
         RTC_DCHECK_EQ(stream.get(), existing_stream.get());
         added = false;
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
index 8b56b8b4f10..367a53780c9 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
@@ -59,12 +59,12 @@ void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
         this, &RtpTransceiver::OnFirstPacketReceived);
   }
 
-  for (auto sender : senders_) {
+  for (const auto& sender : senders_) {
     sender->internal()->SetMediaChannel(channel_ ? channel_->media_channel()
                                                  : nullptr);
   }
 
-  for (auto receiver : receivers_) {
+  for (const auto& receiver : receivers_) {
     if (!channel_) {
       receiver->internal()->Stop();
     }
@@ -147,7 +147,7 @@ absl::optional<std::string> RtpTransceiver::mid() const {
 }
 
 void RtpTransceiver::OnFirstPacketReceived(cricket::ChannelInterface*) {
-  for (auto receiver : receivers_) {
+  for (const auto& receiver : receivers_) {
     receiver->internal()->NotifyFirstPacketReceived();
   }
 }
@@ -212,10 +212,10 @@ absl::optional<RtpTransceiverDirection> RtpTransceiver::fired_direction()
 }
 
 void RtpTransceiver::Stop() {
-  for (auto sender : senders_) {
+  for (const auto& sender : senders_) {
     sender->internal()->Stop();
   }
-  for (auto receiver : receivers_) {
+  for (const auto& receiver : receivers_) {
     receiver->internal()->Stop();
   }
   stopped_ = true;
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
index 3cd85e69b23..3fc988b49eb 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
@@ -871,7 +871,7 @@ void StatsCollector::ExtractBweInfo() {
 
   // Fill in target encoder bitrate, actual encoder bitrate, rtx bitrate, etc.
   // TODO(holmer): Also fill this in for audio.
-  for (auto transceiver : pc_->GetTransceiversInternal()) {
+  for (const auto& transceiver : pc_->GetTransceiversInternal()) {
     if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO) {
       continue;
     }
@@ -912,7 +912,7 @@ void StatsCollector::ExtractMediaInfo() {
 
   {
     rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
-    for (auto transceiver : pc_->GetTransceiversInternal()) {
+    for (const auto& transceiver : pc_->GetTransceiversInternal()) {
       if (!transceiver->internal()->channel()) {
         continue;
       }
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
index b5abb7e9365..51826df45fc 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
@@ -47,7 +47,7 @@ void GetAudioAndVideoTrackBySsrc(
   // means one thread jump if on signaling thread and two thread jumps if on any
   // other threads). Is there a way to avoid thread jump(s) on a per
   // sender/receiver, per method basis?
-  for (auto rtp_sender : rtp_senders) {
+  for (const auto& rtp_sender : rtp_senders) {
     cricket::MediaType media_type = rtp_sender->media_type();
     MediaStreamTrackInterface* track = rtp_sender->track();
     if (!track) {
@@ -72,7 +72,7 @@ void GetAudioAndVideoTrackBySsrc(
       }
     }
   }
-  for (auto rtp_receiver : rtp_receivers) {
+  for (const auto& rtp_receiver : rtp_receivers) {
     cricket::MediaType media_type = rtp_receiver->media_type();
     MediaStreamTrackInterface* track = rtp_receiver->track();
     RTC_DCHECK(track);
@@ -126,10 +126,10 @@ TrackMediaInfoMap::TrackMediaInfoMap(
       &remote_video_track_by_ssrc, &unsignaled_audio_track,
       &unsignaled_video_track);
 
-  for (auto sender : rtp_senders) {
+  for (const auto& sender : rtp_senders) {
     attachment_id_by_track_[sender->track()] = sender->AttachmentId();
   }
-  for (auto receiver : rtp_receivers) {
+  for (const auto& receiver : rtp_receivers) {
     attachment_id_by_track_[receiver->track()] = receiver->AttachmentId();
   }
 
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
index b1dc5ff9981..38d852e53ce 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
@@ -328,7 +328,7 @@ bool FileRotatingStream::GetSize(size_t* size) const {
   RTC_DCHECK(size);
   *size = 0;
   size_t total_size = 0;
-  for (auto file_name : file_names_) {
+  for (const auto& file_name : file_names_) {
     total_size += GetFileSize(file_name).value_or(0);
   }
   *size = total_size;
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc
index 2e37d55deb1..7b82276728b 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc
@@ -22,7 +22,7 @@ OpenSSLSessionCache::OpenSSLSessionCache(SSLMode ssl_mode, SSL_CTX* ssl_ctx)
 }
 
 OpenSSLSessionCache::~OpenSSLSessionCache() {
-  for (auto it : sessions_) {
+  for (const auto& it : sessions_) {
     SSL_SESSION_free(it.second);
   }
   SSL_CTX_free(ssl_ctx_);
diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
index e20b7d294fd..e82e967cf4f 100644
--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
@@ -275,7 +275,7 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   // Aggregate content_specific_stats_ by removing experiment or simulcast
   // information;
   std::map<VideoContentType, ContentSpecificStats> aggregated_stats;
-  for (auto it : content_specific_stats_) {
+  for (const auto& it : content_specific_stats_) {
     // Calculate simulcast specific metrics (".S0" ... ".S2" suffixes).
     VideoContentType content_type = it.first;
     if (videocontenttypehelpers::GetSimulcastId(content_type) > 0) {
@@ -297,7 +297,7 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
     aggregated_stats[content_type].Add(it.second);
   }
 
-  for (auto it : aggregated_stats) {
+  for (const auto& it : aggregated_stats) {
     // For the metric Foo we report the following slices:
     // WebRTC.Video.Foo,
     // WebRTC.Video.Screenshare.Foo,