From 05cf6be7263ce11f4bd5222c9a3a4cc2a1470a01 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Thu, 31 Jan 2019 21:38:12 +0100 Subject: [PATCH] [clang-tidy] Apply performance-move-const-arg fixes. This CL is a manual spin-off of [1], which tried to apply clang-tidy's performance-move-const-arg [1] to the WebRTC codebase. Since there are some wrong fixes to correct, this CL collects all the fixes that could be landed as is. [1] - https://webrtc-review.googlesource.com/c/src/+/120350 [2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html Bug: webrtc:10252 Change-Id: Ic4882213556344e65c66e27415e91ff6f89134d7 Reviewed-on: https://webrtc-review.googlesource.com/c/120814 Reviewed-by: Karl Wiberg Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#26515} --- api/audio_codecs/opus/audio_decoder_opus.cc | 2 +- api/test/loopback_media_transport.cc | 2 +- call/simulated_network.cc | 4 ++-- .../codecs/opus/audio_encoder_opus_unittest.cc | 6 +++--- modules/pacing/paced_sender.cc | 2 +- modules/video_coding/rtp_frame_reference_finder.cc | 4 ++-- p2p/base/dtls_transport.cc | 2 +- pc/channel_unittest.cc | 2 +- pc/jsep_transport.cc | 4 ++-- pc/peer_connection.cc | 4 ++-- pc/rtp_sender_receiver_unittest.cc | 12 ++++++------ pc/rtp_transport_unittest.cc | 2 +- .../single_process_encoded_image_id_injector.cc | 4 ++-- video/video_replay.cc | 2 +- 14 files changed, 26 insertions(+), 26 deletions(-) diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc index 2f1668b825..cd7041681f 100644 --- a/api/audio_codecs/opus/audio_decoder_opus.cc +++ b/api/audio_codecs/opus/audio_decoder_opus.cc @@ -51,7 +51,7 @@ void AudioDecoderOpus::AppendSupportedDecoders( opus_info.supports_network_adaption = true; SdpAudioFormat opus_format( {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}); - specs->push_back({std::move(opus_format), std::move(opus_info)}); + specs->push_back({std::move(opus_format), opus_info}); } std::unique_ptr AudioDecoderOpus::MakeAudioDecoder( diff --git a/api/test/loopback_media_transport.cc b/api/test/loopback_media_transport.cc index e7ccb0afd8..5b75e4048d 100644 --- a/api/test/loopback_media_transport.cc +++ b/api/test/loopback_media_transport.cc @@ -124,7 +124,7 @@ RTCError MediaTransportPair::LoopbackMediaTransport::SendAudioFrame( ++stats_.sent_audio_frames; } invoker_.AsyncInvoke(RTC_FROM_HERE, thread_, [this, channel_id, frame] { - other_->OnData(channel_id, std::move(frame)); + other_->OnData(channel_id, frame); }); return RTCError::OK(); } diff --git a/call/simulated_network.cc b/call/simulated_network.cc index 9bb8bab360..0884b295f8 100644 --- a/call/simulated_network.cc +++ b/call/simulated_network.cc @@ -125,7 +125,7 @@ void SimulatedNetwork::UpdateCapacityQueue(ConfigState state, } // Time to get this packet. - PacketInfo packet = std::move(capacity_link_.front()); + PacketInfo packet = capacity_link_.front(); capacity_link_.pop(); time_us += time_until_front_exits_us; @@ -165,7 +165,7 @@ void SimulatedNetwork::UpdateCapacityQueue(ConfigState state, needs_sort = true; } } - delay_link_.emplace_back(std::move(packet)); + delay_link_.emplace_back(packet); } last_capacity_link_visit_us_ = time_now_us; // Cannot save unused capacity for later. diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc index 6b34361813..b3b531fca3 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc @@ -82,9 +82,9 @@ std::unique_ptr CreateCodec(size_t num_channels) { new MockSmoothingFilter()); states->mock_bitrate_smoother = bitrate_smoother.get(); - states->encoder.reset(new AudioEncoderOpusImpl( - states->config, kDefaultOpusPayloadType, std::move(creator), - std::move(bitrate_smoother))); + states->encoder.reset( + new AudioEncoderOpusImpl(states->config, kDefaultOpusPayloadType, creator, + std::move(bitrate_smoother))); return states; } diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc index a30aef059e..2a10b427a2 100644 --- a/modules/pacing/paced_sender.cc +++ b/modules/pacing/paced_sender.cc @@ -359,7 +359,7 @@ void PacedSender::Process() { if (success) { bytes_sent += packet->bytes; // Send succeeded, remove it from the queue. - OnPacketSent(std::move(packet)); + OnPacketSent(packet); if (is_probing && bytes_sent > recommended_probe_size) break; } else { diff --git a/modules/video_coding/rtp_frame_reference_finder.cc b/modules/video_coding/rtp_frame_reference_finder.cc index 45f710ae07..6d4860d4b9 100644 --- a/modules/video_coding/rtp_frame_reference_finder.cc +++ b/modules/video_coding/rtp_frame_reference_finder.cc @@ -275,7 +275,7 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8( if (codec_header.pictureId == kNoPictureId || codec_header.temporalIdx == kNoTemporalIdx || codec_header.tl0PicIdx == kNoTl0PicIdx) { - return ManageFramePidOrSeqNum(std::move(frame), codec_header.pictureId); + return ManageFramePidOrSeqNum(frame, codec_header.pictureId); } frame->id.picture_id = codec_header.pictureId % kPicIdLength; @@ -424,7 +424,7 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp9( if (codec_header.picture_id == kNoPictureId || codec_header.temporal_idx == kNoTemporalIdx) { - return ManageFramePidOrSeqNum(std::move(frame), codec_header.picture_id); + return ManageFramePidOrSeqNum(frame, codec_header.picture_id); } frame->id.spatial_layer = codec_header.spatial_idx; diff --git a/p2p/base/dtls_transport.cc b/p2p/base/dtls_transport.cc index ba565ca96f..d3db35bfb9 100644 --- a/p2p/base/dtls_transport.cc +++ b/p2p/base/dtls_transport.cc @@ -208,7 +208,7 @@ bool DtlsTransport::SetDtlsRole(rtc::SSLRole role) { return true; } - dtls_role_ = std::move(role); + dtls_role_ = role; return true; } diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc index 6e3aa194f2..edcea883bf 100644 --- a/pc/channel_unittest.cc +++ b/pc/channel_unittest.cc @@ -1381,7 +1381,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> { webrtc::RtpParameters BitrateLimitedParameters(absl::optional limit) { webrtc::RtpParameters parameters; webrtc::RtpEncodingParameters encoding; - encoding.max_bitrate_bps = std::move(limit); + encoding.max_bitrate_bps = limit; parameters.encodings.push_back(encoding); return parameters; } diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc index fd6bd0de23..28c752575d 100644 --- a/pc/jsep_transport.cc +++ b/pc/jsep_transport.cc @@ -651,8 +651,8 @@ webrtc::RTCError JsepTransport::NegotiateDtlsRole( // If local is passive, local will act as server. } - *negotiated_dtls_role = (is_remote_server ? std::move(rtc::SSL_CLIENT) - : std::move(rtc::SSL_SERVER)); + *negotiated_dtls_role = + (is_remote_server ? rtc::SSL_CLIENT : rtc::SSL_SERVER); return webrtc::RTCError::OK(); } diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index e0479d134a..770c4abac0 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -3739,7 +3739,7 @@ void PeerConnection::CreateAudioReceiver( auto receiver = RtpReceiverProxyWithInternal::Create( signaling_thread(), audio_receiver); GetAudioTransceiver()->internal()->AddReceiver(receiver); - Observer()->OnAddTrack(receiver, std::move(streams)); + Observer()->OnAddTrack(receiver, streams); NoteUsageEvent(UsageEvent::AUDIO_ADDED); } @@ -3757,7 +3757,7 @@ void PeerConnection::CreateVideoReceiver( auto receiver = RtpReceiverProxyWithInternal::Create( signaling_thread(), video_receiver); GetVideoTransceiver()->internal()->AddReceiver(receiver); - Observer()->OnAddTrack(receiver, std::move(streams)); + Observer()->OnAddTrack(receiver, streams); NoteUsageEvent(UsageEvent::VIDEO_ADDED); } diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc index 83f06c6449..ba0e0b5e06 100644 --- a/pc/rtp_sender_receiver_unittest.cc +++ b/pc/rtp_sender_receiver_unittest.cc @@ -239,8 +239,8 @@ class RtpSenderReceiverTest : public testing::Test, void CreateAudioRtpReceiver( std::vector> streams = {}) { - audio_rtp_receiver_ = new AudioRtpReceiver( - rtc::Thread::Current(), kAudioTrackId, std::move(streams)); + audio_rtp_receiver_ = + new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams); audio_rtp_receiver_->SetMediaChannel(voice_media_channel_); audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc); audio_track_ = audio_rtp_receiver_->audio_track(); @@ -249,8 +249,8 @@ class RtpSenderReceiverTest : public testing::Test, void CreateVideoRtpReceiver( std::vector> streams = {}) { - video_rtp_receiver_ = new VideoRtpReceiver( - rtc::Thread::Current(), kVideoTrackId, std::move(streams)); + video_rtp_receiver_ = + new VideoRtpReceiver(rtc::Thread::Current(), kVideoTrackId, streams); video_rtp_receiver_->SetMediaChannel(video_media_channel_); video_rtp_receiver_->SetupMediaChannel(kVideoSsrc); video_track_ = video_rtp_receiver_->video_track(); @@ -269,8 +269,8 @@ class RtpSenderReceiverTest : public testing::Test, video_media_channel_->AddRecvStream(stream_params); uint32_t primary_ssrc = stream_params.first_ssrc(); - video_rtp_receiver_ = new VideoRtpReceiver( - rtc::Thread::Current(), kVideoTrackId, std::move(streams)); + video_rtp_receiver_ = + new VideoRtpReceiver(rtc::Thread::Current(), kVideoTrackId, streams); video_rtp_receiver_->SetMediaChannel(video_media_channel_); video_rtp_receiver_->SetupMediaChannel(primary_ssrc); video_track_ = video_rtp_receiver_->video_track(); diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc index 1079ab46d6..f617445a06 100644 --- a/pc/rtp_transport_unittest.cc +++ b/pc/rtp_transport_unittest.cc @@ -86,7 +86,7 @@ class SignalObserver : public sigslot::has_slots<> { absl::optional network_route() { return network_route_; } void OnNetworkRouteChanged(absl::optional network_route) { - network_route_ = std::move(network_route); + network_route_ = network_route; } void OnSentPacket(rtc::PacketTransportInternal* packet_transport, diff --git a/test/pc/e2e/analyzer/video/single_process_encoded_image_id_injector.cc b/test/pc/e2e/analyzer/video/single_process_encoded_image_id_injector.cc index 0d72fd90f5..105ee8eacd 100644 --- a/test/pc/e2e/analyzer/video/single_process_encoded_image_id_injector.cc +++ b/test/pc/e2e/analyzer/video/single_process_encoded_image_id_injector.cc @@ -45,7 +45,7 @@ EncodedImage SingleProcessEncodedImageIdInjector::InjectId( // Will create new one if missed. ExtractionInfoVector& ev = extraction_cache_[id]; info.sub_id = ev.next_sub_id++; - ev.infos[info.sub_id] = std::move(info); + ev.infos[info.sub_id] = info; } EncodedImage out = source; @@ -83,7 +83,7 @@ EncodedImageWithId SingleProcessEncodedImageIdInjector::ExtractId( auto info_it = ext_vector_it->second.infos.find(sub_id); RTC_CHECK(info_it != ext_vector_it->second.infos.end()) << "Unknown sub id " << sub_id << " for frame " << next_id; - info = std::move(info_it->second); + info = info_it->second; ext_vector_it->second.infos.erase(info_it); } diff --git a/video/video_replay.cc b/video/video_replay.cc index 071df8ce90..6cfb8c1b71 100644 --- a/video/video_replay.cc +++ b/video/video_replay.cc @@ -247,7 +247,7 @@ class RtpReplayer final { const std::string& rtp_dump_path) { webrtc::RtcEventLogNullImpl event_log; Call::Config call_config(&event_log); - std::unique_ptr call(Call::Create(std::move(call_config))); + std::unique_ptr call(Call::Create(call_config)); std::unique_ptr stream_state; // Attempt to load the configuration if (replay_config_path.empty()) {