[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 <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26515}diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc
index 2f1668b..cd70416 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -51,7 +51,7 @@
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<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
diff --git a/api/test/loopback_media_transport.cc b/api/test/loopback_media_transport.cc
index e7ccb0a..5b75e40 100644
--- a/api/test/loopback_media_transport.cc
+++ b/api/test/loopback_media_transport.cc
@@ -124,7 +124,7 @@
++stats_.sent_audio_frames;
}
invoker_.AsyncInvoke<void>(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 9bb8bab..0884b29 100644
--- a/call/simulated_network.cc
+++ b/call/simulated_network.cc
@@ -125,7 +125,7 @@
}
// 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 @@
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 6b34361..b3b531f 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 @@
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 a30aef0..2a10b42 100644
--- a/modules/pacing/paced_sender.cc
+++ b/modules/pacing/paced_sender.cc
@@ -359,7 +359,7 @@
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 45f710a..6d4860d 100644
--- a/modules/video_coding/rtp_frame_reference_finder.cc
+++ b/modules/video_coding/rtp_frame_reference_finder.cc
@@ -275,7 +275,7 @@
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 @@
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 ba565ca..d3db35b 100644
--- a/p2p/base/dtls_transport.cc
+++ b/p2p/base/dtls_transport.cc
@@ -208,7 +208,7 @@
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 6e3aa19..edcea88 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -1381,7 +1381,7 @@
webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> 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 fd6bd0d..28c7525 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -651,8 +651,8 @@
// 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 e0479d1..770c4ab 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -3739,7 +3739,7 @@
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::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 @@
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::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 83f06c6..ba0e0b5 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -239,8 +239,8 @@
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> 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 @@
void CreateVideoRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> 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 @@
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 1079ab4..f617445 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -86,7 +86,7 @@
absl::optional<rtc::NetworkRoute> network_route() { return network_route_; }
void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> 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 0d72fd9..105ee8e 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 @@
// 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 @@
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 071df8c..6cfb8c1 100644
--- a/video/video_replay.cc
+++ b/video/video_replay.cc
@@ -247,7 +247,7 @@
const std::string& rtp_dump_path) {
webrtc::RtcEventLogNullImpl event_log;
Call::Config call_config(&event_log);
- std::unique_ptr<Call> call(Call::Create(std::move(call_config)));
+ std::unique_ptr<Call> call(Call::Create(call_config));
std::unique_ptr<StreamState> stream_state;
// Attempt to load the configuration
if (replay_config_path.empty()) {
