Attachment 359283 Details for Bug 193498 – Patch v1

[patch] Patch v1

bug-193498-20190116111949.patch (text/plain), 66.77 KB, created by David Kilzer (:ddkilzer) on 2019-01-16 11:19:50 PST

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Creator: David Kilzer (:ddkilzer)

Created: 2019-01-16 11:19:50 PST

Size: 66.77 KB

patch

obsolete

>Subversion Revision: 240009
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index c71bf115911fee3101cef7adf95c4a71d61392d8..dd749ee30e24eba7d23b15934b19c71d9f8ddeab 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,30 @@
>+2019-01-16  David Kilzer  <ddkilzer@apple.com>
>+
>+        clang-tidy: Fix unnecessary copy/ref churn of for loop variables in libwebrtc
>+        <https://webkit.org/b/193498>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Fix unwanted copying/ref churn of loop variables by making them
>+        const references.
>+
>+        * Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc:
>+        * Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc:
>+        * Source/webrtc/p2p/base/mdns_message.cc:
>+        * Source/webrtc/p2p/base/port.cc:
>+        * Source/webrtc/p2p/base/stunrequest.cc:
>+        * Source/webrtc/pc/jseptransportcontroller.cc:
>+        * Source/webrtc/pc/peerconnection.cc:
>+        * Source/webrtc/pc/rtcstatscollector.cc:
>+        * Source/webrtc/pc/rtpreceiver.cc:
>+        * Source/webrtc/pc/rtptransceiver.cc:
>+        * Source/webrtc/pc/statscollector.cc:
>+        * Source/webrtc/pc/trackmediainfomap.cc:
>+        * Source/webrtc/rtc_base/filerotatingstream.cc:
>+        * Source/webrtc/rtc_base/opensslsessioncache.cc:
>+        * Source/webrtc/video/receive_statistics_proxy.cc:
>+        * WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff: Added.
>+
> 2019-01-15  David Kilzer  <ddkilzer@apple.com>
> 
>         REGRESSION (r239510): Remove duplicate copy of srtpsession.cc from 'webrtcpcrtc' target in Xcode project
>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 5d7f8aa2c36d7fd66d562aacb86de6ce7666ce47..7a82666e81203286c9cf0a18f04c87fdd06bfd9a 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 383f785dfea82062251d0a4fca71e6fab9a278a7..08405046528c57d209ff5d2ebaa4d15aa0622db8 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 f14a0d117e35ea39c221ee1fff0cf7823d994f81..321d21d35bd58234c27b4d5b8eeb092cb43604c2 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 bb20a4d5da39ed90e24ddbc41cb3db46c445b6b9..a180180b91d6a75c3a1e0321891127910e49b476 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 edba4d68e54bc20a0ef3fd466a84e5c9d3632fce..dfb27fccb177d8a4e47cc3de672ca87eadd4d05d 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 78ecaf31deac4e41e8db488a0df6ca81504d2cb7..328e3ca42b18fad29d08678c1c2d0120231391bd 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 4f0a8ea01127622720d0cbddc2e1e17cc940225c..ee8a909e03e86fdfc30b239ccb540123a14a3588 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 d48ecc01f359e51b900259828c085c8b5e1617f2..6fc4118d979946d1b824c926c039fa6fb9d3a4af 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 1916a7367032c4bb1f81a6510d2eef11c45a9279..e5a4373294cd25677a89fca61188f30eba5cec5c 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 8b56b8b4f10a529f9b9a3b3e0814214d1664b769..367a53780c9cf43968911fbba6965c4022ff42a7 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 3cd85e69b23bf36513060debaad39c71434b85f2..3fc988b49eb6099f8ae2d2ecdcfc0669582c3cc3 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 b5abb7e9365d7fed81b55569e623b6c2040b8ab8..51826df45fc77c48062d7e0e7f29423b063cac70 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 b1dc5ff99817851c1c9521b4e087b02abe509369..38d852e53ce72ed1c3bc85edfe953bab2f693b49 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 2e37d55deb1a5ff25757b3991c90f576a357c243..7b82276728b7236d421f7a833317af7d54b11d54 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 e20b7d294fd6224942c7798ae66403b5d8357ddd..e82e967cf4fc29bae431c038fa125023a3961a03 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,
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff b/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff
>new file mode 100644
>index 0000000000000000000000000000000000000000..c2a1e245a9d7b31658f8175fc30a2e90fd7fa680
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff
>@@ -0,0 +1,664 @@
>+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,

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Attachments on bug 193498: 359283