| 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, |
