Diff
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2019-01-16 22:44:54 UTC (rev 240053)
@@ -1,3 +1,30 @@
+2019-01-16 David Kilzer <ddkil...@apple.com>
+
+ clang-tidy: Fix unnecessary copy/ref churn of for loop variables in libwebrtc
+ <https://webkit.org/b/193498>
+
+ Reviewed by Youenn Fablet.
+
+ 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 <ddkil...@apple.com>
REGRESSION (r239510): Remove duplicate copy of srtpsession.cc from 'webrtcpcrtc' target in Xcode project
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -124,7 +124,7 @@
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();
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -437,7 +437,7 @@
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);
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -353,7 +353,7 @@
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 @@
return true;
};
- for (auto question : question_section_) {
+ for (const auto& question : question_section_) {
if (!question.Write(buf)) {
return false;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -948,7 +948,7 @@
// 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);
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -69,7 +69,7 @@
}
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 @@
}
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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -629,7 +629,7 @@
// 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 @@
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 @@
}
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 @@
// 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 @@
// 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> 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;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -832,7 +832,7 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
}
}
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 @@
}
if (IsUnifiedPlan()) {
- for (auto transceiver : transceivers_) {
+ for (const auto& transceiver : transceivers_) {
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
@@ -2447,7 +2447,7 @@
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 @@
}
// 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 @@
// 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 @@
ChangeSignalingState(PeerConnectionInterface::kClosed);
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
- for (auto transceiver : transceivers_) {
+ for (const auto& transceiver : transceivers_) {
transceiver->Stop();
}
@@ -4045,7 +4045,7 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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::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::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();
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -534,7 +534,7 @@
// 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 @@
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 @@
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 @@
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());
@@ -1150,7 +1150,7 @@
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,
@@ -1157,7 +1157,7 @@
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 @@
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 @@
}
}
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>(
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -207,9 +207,9 @@
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 @@
}
}
// 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::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 @@
}
}
// 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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -59,12 +59,12 @@
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 @@
}
void RtpTransceiver::OnFirstPacketReceived(cricket::ChannelInterface*) {
- for (auto receiver : receivers_) {
+ for (const auto& receiver : receivers_) {
receiver->internal()->NotifyFirstPacketReceived();
}
}
@@ -212,10 +212,10 @@
}
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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -871,7 +871,7 @@
// 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 @@
{
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
- for (auto transceiver : pc_->GetTransceiversInternal()) {
+ for (const auto& transceiver : pc_->GetTransceiversInternal()) {
if (!transceiver->internal()->channel()) {
continue;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -47,7 +47,7 @@
// 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 @@
}
}
}
- 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 @@
&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();
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -328,7 +328,7 @@
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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -22,7 +22,7 @@
}
OpenSSLSessionCache::~OpenSSLSessionCache() {
- for (auto it : sessions_) {
+ for (const auto& it : sessions_) {
SSL_SESSION_free(it.second);
}
SSL_CTX_free(ssl_ctx_);
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc (240052 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc 2019-01-16 22:08:46 UTC (rev 240052)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc 2019-01-16 22:44:54 UTC (rev 240053)
@@ -275,7 +275,7 @@
// 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 @@
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,
Added: trunk/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff (0 => 240053)
--- trunk/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/WebKit/0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff 2019-01-16 22:44:54 UTC (rev 240053)
@@ -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,
