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Fixing some clang-tidy findings.

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Bug: None
Change-Id: I949c1ff35284ce79c99e8f76148f63b8bba965a9
Reviewed-on: https://webrtc-review.googlesource.com/24041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20818}
This commit is contained in:
Mirko Bonadei 2017-11-21 17:04:20 +01:00 committed by Commit Bot
parent 89e712649d
commit c61ce0d0cd
11 changed files with 39 additions and 25 deletions
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2
media/base/codec_unittest.cc
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@ -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) {}

3
pc/channel_unittest.cc
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@ -9,6 +9,7 @@
*/
#include <memory>
#include <utility>
#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<int> 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;
}

15
pc/peerconnection_integrationtest.cc
View File

@ -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<void(cricket::SessionDescription*)> 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<void(cricket::SessionDescription*)> 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<webrtc::AudioTrackInterface> audio,
rtc::scoped_refptr<webrtc::VideoTrackInterface> video) {
const rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio,
const rtc::scoped_refptr<webrtc::VideoTrackInterface>& video) {
rtc::scoped_refptr<MediaStreamInterface> 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<RtpReceiverInterface>& receiver :
pc()->GetReceivers()) {
std::unique_ptr<MockRtpReceiverObserver> observer(
new MockRtpReceiverObserver(receiver->media_type()));
receiver->SetObserver(observer.get());
@ -723,7 +726,7 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
}
std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer) {
MockCreateSessionDescriptionObserver* observer) {
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (!observer->result()) {
return nullptr;

15
pc/peerconnectionendtoend_unittest.cc
View File

@ -73,10 +73,11 @@ class PeerConnectionEndToEndTest
#endif
}
void CreatePcs(
const MediaConstraintsInterface* pc_constraints,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
void CreatePcs(const MediaConstraintsInterface* pc_constraints,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
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,

5
pc/peerconnectioninterface_unittest.cc
View File

@ -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<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();

2
pc/rtcstats_integrationtest.cc
View File

@ -267,7 +267,7 @@ class RTCStatsVerifier {
valid_reference = true;
const RTCStatsMember<std::vector<std::string>>& ids =
member.cast_to<RTCStatsMember<std::vector<std::string>>>();
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;

6
pc/rtcstatscollector_unittest.cc
View File

@ -235,7 +235,8 @@ rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
}
rtc::scoped_refptr<MockRtpSender> CreateMockSender(
rtc::scoped_refptr<MediaStreamTrackInterface> track, uint32_t ssrc) {
const rtc::scoped_refptr<MediaStreamTrackInterface>& track,
uint32_t ssrc) {
rtc::scoped_refptr<MockRtpSender> sender(
new rtc::RefCountedObject<MockRtpSender>());
EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
@ -254,7 +255,8 @@ rtc::scoped_refptr<MockRtpSender> CreateMockSender(
}
rtc::scoped_refptr<MockRtpReceiver> CreateMockReceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track, uint32_t ssrc) {
const rtc::scoped_refptr<MediaStreamTrackInterface>& track,
uint32_t ssrc) {
rtc::scoped_refptr<MockRtpReceiver> receiver(
new rtc::RefCountedObject<MockRtpReceiver>());
EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track));

3
pc/rtpsenderreceiver_unittest.cc
View File

@ -129,7 +129,8 @@ class RtpSenderReceiverTest : public testing::Test,
void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
void CreateAudioRtpSender(
const rtc::scoped_refptr<LocalAudioSource>& source) {
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
audio_rtp_sender_ =

3
pc/rtptransport_unittest.cc
View File

@ -9,6 +9,7 @@
*/
#include <string>
#include <utility>
#include "p2p/base/fakepackettransport.h"
#include "pc/rtptransport.h"
@ -69,7 +70,7 @@ class SignalObserver : public sigslot::has_slots<> {
rtc::Optional<rtc::NetworkRoute> network_route() { return network_route_; }
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
network_route_ = network_route;
network_route_ = std::move(network_route);
}
private:

4
pc/statscollector_unittest.cc
View File

@ -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);

6
pc/trackmediainfomap_unittest.cc
View File

@ -53,7 +53,8 @@ rtc::scoped_refptr<MockRtpSender> CreateMockRtpSender(
}
rtc::scoped_refptr<MockRtpSender> sender(
new rtc::RefCountedObject<MockRtpSender>());
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<MockRtpReceiver> CreateMockRtpReceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
rtc::scoped_refptr<MockRtpReceiver> receiver(
new rtc::RefCountedObject<MockRtpReceiver>());
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())