From c61ce0d0cd03c912e108f261c096214b917f8f72 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 21 Nov 2017 17:04:20 +0100 Subject: [PATCH] Fixing some clang-tidy findings. Bug: None Change-Id: I949c1ff35284ce79c99e8f76148f63b8bba965a9 Reviewed-on: https://webrtc-review.googlesource.com/24041 Commit-Queue: Mirko Bonadei Reviewed-by: Karl Wiberg Cr-Commit-Position: refs/heads/master@{#20818} --- media/base/codec_unittest.cc | 2 +- pc/channel_unittest.cc | 3 ++- pc/peerconnection_integrationtest.cc | 15 +++++++++------ pc/peerconnectionendtoend_unittest.cc | 15 +++++++++------ pc/peerconnectioninterface_unittest.cc | 5 +++-- pc/rtcstats_integrationtest.cc | 2 +- pc/rtcstatscollector_unittest.cc | 6 ++++-- pc/rtpsenderreceiver_unittest.cc | 3 ++- pc/rtptransport_unittest.cc | 3 ++- pc/statscollector_unittest.cc | 4 ++-- pc/trackmediainfomap_unittest.cc | 6 ++++-- 11 files changed, 39 insertions(+), 25 deletions(-) diff --git a/media/base/codec_unittest.cc b/media/base/codec_unittest.cc index 00d23eec80..03d8684c64 100644 --- a/media/base/codec_unittest.cc +++ b/media/base/codec_unittest.cc @@ -22,7 +22,7 @@ using cricket::kCodecParamMinBitrate; class TestCodec : public Codec { public: - TestCodec(int id, const std::string name, int clockrate) + TestCodec(int id, const std::string& name, int clockrate) : Codec(id, name, clockrate) {} TestCodec() : Codec() {} TestCodec(const TestCodec& c) : Codec(c) {} diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc index 52cb419495..36a6b57d90 100644 --- a/pc/channel_unittest.cc +++ b/pc/channel_unittest.cc @@ -9,6 +9,7 @@ */ #include +#include #include "api/array_view.h" #include "media/base/fakemediaengine.h" @@ -1848,7 +1849,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> { webrtc::RtpParameters BitrateLimitedParameters(rtc::Optional limit) { webrtc::RtpParameters parameters; webrtc::RtpEncodingParameters encoding; - encoding.max_bitrate_bps = limit; + encoding.max_bitrate_bps = std::move(limit); parameters.encodings.push_back(encoding); return parameters; } diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc index 90aad09078..8ce692e64e 100644 --- a/pc/peerconnection_integrationtest.cc +++ b/pc/peerconnection_integrationtest.cc @@ -28,6 +28,7 @@ #include "api/mediastreaminterface.h" #include "api/peerconnectioninterface.h" #include "api/peerconnectionproxy.h" +#include "api/rtpreceiverinterface.h" #include "api/test/fakeconstraints.h" #include "media/engine/fakewebrtcvideoengine.h" #include "p2p/base/p2pconstants.h" @@ -80,6 +81,7 @@ using webrtc::PeerConnection; using webrtc::PeerConnectionInterface; using webrtc::PeerConnectionFactory; using webrtc::PeerConnectionProxy; +using webrtc::RtpReceiverInterface; using webrtc::SessionDescriptionInterface; using webrtc::StreamCollectionInterface; @@ -276,14 +278,14 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, // generate, but a non-JSEP endpoint might. void SetReceivedSdpMunger( std::function munger) { - received_sdp_munger_ = munger; + received_sdp_munger_ = std::move(munger); } // Similar to the above, but this is run on SDP immediately after it's // generated. void SetGeneratedSdpMunger( std::function munger) { - generated_sdp_munger_ = munger; + generated_sdp_munger_ = std::move(munger); } // Every ICE connection state in order that has been seen by the observer. @@ -343,8 +345,8 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, } void AddMediaStreamFromTracks( - rtc::scoped_refptr audio, - rtc::scoped_refptr video) { + const rtc::scoped_refptr& audio, + const rtc::scoped_refptr& video) { rtc::scoped_refptr stream = peer_connection_factory_->CreateLocalMediaStream( rtc::CreateRandomUuid()); @@ -553,7 +555,8 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, void ResetRtpReceiverObservers() { rtp_receiver_observers_.clear(); - for (auto receiver : pc()->GetReceivers()) { + for (const rtc::scoped_refptr& receiver : + pc()->GetReceivers()) { std::unique_ptr observer( new MockRtpReceiverObserver(receiver->media_type())); receiver->SetObserver(observer.get()); @@ -723,7 +726,7 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, } std::unique_ptr WaitForDescriptionFromObserver( - rtc::scoped_refptr observer) { + MockCreateSessionDescriptionObserver* observer) { EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout); if (!observer->result()) { return nullptr; diff --git a/pc/peerconnectionendtoend_unittest.cc b/pc/peerconnectionendtoend_unittest.cc index dce69dabcc..8b16610419 100644 --- a/pc/peerconnectionendtoend_unittest.cc +++ b/pc/peerconnectionendtoend_unittest.cc @@ -73,10 +73,11 @@ class PeerConnectionEndToEndTest #endif } - void CreatePcs( - const MediaConstraintsInterface* pc_constraints, - rtc::scoped_refptr audio_encoder_factory, - rtc::scoped_refptr audio_decoder_factory) { + void CreatePcs(const MediaConstraintsInterface* pc_constraints, + const rtc::scoped_refptr& + audio_encoder_factory, + const rtc::scoped_refptr& + audio_decoder_factory) { EXPECT_TRUE(caller_->CreatePc( pc_constraints, config_, audio_encoder_factory, audio_decoder_factory)); EXPECT_TRUE(callee_->CreatePc( @@ -95,8 +96,10 @@ class PeerConnectionEndToEndTest GetAndAddUserMedia(true, audio_constraints, true, video_constraints); } - void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints, - bool video, FakeConstraints video_constraints) { + void GetAndAddUserMedia(bool audio, + const FakeConstraints& audio_constraints, + bool video, + const FakeConstraints& video_constraints) { caller_->GetAndAddUserMedia(audio, audio_constraints, video, video_constraints); callee_->GetAndAddUserMedia(audio, audio_constraints, diff --git a/pc/peerconnectioninterface_unittest.cc b/pc/peerconnectioninterface_unittest.cc index 409fa91f62..389aaeb596 100644 --- a/pc/peerconnectioninterface_unittest.cc +++ b/pc/peerconnectioninterface_unittest.cc @@ -653,8 +653,9 @@ class PeerConnectionInterfaceTest : public testing::Test { CreatePeerConnection(config, nullptr); } - void CreatePeerConnection(PeerConnectionInterface::RTCConfiguration config, - webrtc::MediaConstraintsInterface* constraints) { + void CreatePeerConnection( + const PeerConnectionInterface::RTCConfiguration& config, + webrtc::MediaConstraintsInterface* constraints) { std::unique_ptr port_allocator( new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr)); port_allocator_ = port_allocator.get(); diff --git a/pc/rtcstats_integrationtest.cc b/pc/rtcstats_integrationtest.cc index 3da108300e..d0a5863f65 100644 --- a/pc/rtcstats_integrationtest.cc +++ b/pc/rtcstats_integrationtest.cc @@ -267,7 +267,7 @@ class RTCStatsVerifier { valid_reference = true; const RTCStatsMember>& ids = member.cast_to>>(); - for (const std::string id : *ids) { + for (const std::string& id : *ids) { const RTCStats* referenced_stats = report_->Get(id); if (!referenced_stats || referenced_stats->type() != expected_type) { valid_reference = false; diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc index 4b24214338..d088221051 100644 --- a/pc/rtcstatscollector_unittest.cc +++ b/pc/rtcstatscollector_unittest.cc @@ -235,7 +235,8 @@ rtc::scoped_refptr CreateFakeTrack( } rtc::scoped_refptr CreateMockSender( - rtc::scoped_refptr track, uint32_t ssrc) { + const rtc::scoped_refptr& track, + uint32_t ssrc) { rtc::scoped_refptr sender( new rtc::RefCountedObject()); EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track)); @@ -254,7 +255,8 @@ rtc::scoped_refptr CreateMockSender( } rtc::scoped_refptr CreateMockReceiver( - rtc::scoped_refptr track, uint32_t ssrc) { + const rtc::scoped_refptr& track, + uint32_t ssrc) { rtc::scoped_refptr receiver( new rtc::RefCountedObject()); EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track)); diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc index f40b494e7e..7ea4d4c712 100644 --- a/pc/rtpsenderreceiver_unittest.cc +++ b/pc/rtpsenderreceiver_unittest.cc @@ -129,7 +129,8 @@ class RtpSenderReceiverTest : public testing::Test, void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); } - void CreateAudioRtpSender(rtc::scoped_refptr source) { + void CreateAudioRtpSender( + const rtc::scoped_refptr& source) { audio_track_ = AudioTrack::Create(kAudioTrackId, source); EXPECT_TRUE(local_stream_->AddTrack(audio_track_)); audio_rtp_sender_ = diff --git a/pc/rtptransport_unittest.cc b/pc/rtptransport_unittest.cc index d6eb336309..3876aa3998 100644 --- a/pc/rtptransport_unittest.cc +++ b/pc/rtptransport_unittest.cc @@ -9,6 +9,7 @@ */ #include +#include #include "p2p/base/fakepackettransport.h" #include "pc/rtptransport.h" @@ -69,7 +70,7 @@ class SignalObserver : public sigslot::has_slots<> { rtc::Optional network_route() { return network_route_; } void OnNetworkRouteChanged(rtc::Optional network_route) { - network_route_ = network_route; + network_route_ = std::move(network_route); } private: diff --git a/pc/statscollector_unittest.cc b/pc/statscollector_unittest.cc index 2211f3fb54..d5bc0ff4a8 100644 --- a/pc/statscollector_unittest.cc +++ b/pc/statscollector_unittest.cc @@ -242,12 +242,12 @@ const StatsReport* FindReportById(const StatsReports& reports, return nullptr; } -std::string ExtractSsrcStatsValue(StatsReports reports, +std::string ExtractSsrcStatsValue(const StatsReports& reports, StatsReport::StatsValueName name) { return ExtractStatsValue(StatsReport::kStatsReportTypeSsrc, reports, name); } -std::string ExtractBweStatsValue(StatsReports reports, +std::string ExtractBweStatsValue(const StatsReports& reports, StatsReport::StatsValueName name) { return ExtractStatsValue( StatsReport::kStatsReportTypeBwe, reports, name); diff --git a/pc/trackmediainfomap_unittest.cc b/pc/trackmediainfomap_unittest.cc index 4f71d3b4c2..9f25646556 100644 --- a/pc/trackmediainfomap_unittest.cc +++ b/pc/trackmediainfomap_unittest.cc @@ -53,7 +53,8 @@ rtc::scoped_refptr CreateMockRtpSender( } rtc::scoped_refptr sender( new rtc::RefCountedObject()); - EXPECT_CALL(*sender, track()).WillRepeatedly(testing::Return(track)); + EXPECT_CALL(*sender, track()) + .WillRepeatedly(testing::Return(std::move(track))); EXPECT_CALL(*sender, ssrc()).WillRepeatedly(testing::Return(first_ssrc)); EXPECT_CALL(*sender, media_type()) .WillRepeatedly(testing::Return(media_type)); @@ -68,7 +69,8 @@ rtc::scoped_refptr CreateMockRtpReceiver( rtc::scoped_refptr track) { rtc::scoped_refptr receiver( new rtc::RefCountedObject()); - EXPECT_CALL(*receiver, track()).WillRepeatedly(testing::Return(track)); + EXPECT_CALL(*receiver, track()) + .WillRepeatedly(testing::Return(std::move(track))); EXPECT_CALL(*receiver, media_type()) .WillRepeatedly(testing::Return(media_type)); EXPECT_CALL(*receiver, GetParameters())