Using fully qualified names for using declarations.
Using declarations should use fully qualified names (with leading `::`) unless they are referring to a name inside the current namespace. Source: https://abseil.io/tips/119. This CL removes a lot of "using webrtc::*" adding a namespace to the tests. It also removes some unneeded "using" declarations. Bug: webrtc:9855 Change-Id: Id6eb843e9dcee2e458b1ffd0c499df390fa9c45d Reviewed-on: https://webrtc-review.googlesource.com/c/114001 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25987}
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@ -68,55 +68,20 @@
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#include "system_wrappers/include/metrics.h"
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#include "test/gmock.h"
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using cricket::ContentInfo;
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using cricket::FakeWebRtcVideoDecoder;
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using cricket::FakeWebRtcVideoDecoderFactory;
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using cricket::FakeWebRtcVideoEncoder;
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using cricket::FakeWebRtcVideoEncoderFactory;
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using cricket::MediaContentDescription;
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using cricket::StreamParams;
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using rtc::SocketAddress;
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namespace webrtc {
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namespace {
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using ::cricket::ContentInfo;
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using ::cricket::StreamParams;
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using ::rtc::SocketAddress;
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using ::testing::_;
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using ::testing::Combine;
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using ::testing::ElementsAre;
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using ::testing::UnorderedElementsAreArray;
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using ::testing::Return;
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using ::testing::SetArgPointee;
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using ::testing::Values;
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using ::testing::UnorderedElementsAreArray;
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using ::testing::_;
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using webrtc::DataBuffer;
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using webrtc::DataChannelInterface;
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using webrtc::DtmfSender;
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using webrtc::DtmfSenderInterface;
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using webrtc::DtmfSenderObserverInterface;
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using webrtc::FakeVideoTrackRenderer;
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using webrtc::MediaStreamInterface;
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using webrtc::MediaStreamTrackInterface;
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using webrtc::MockCreateSessionDescriptionObserver;
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using webrtc::MockDataChannelObserver;
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using webrtc::MockSetSessionDescriptionObserver;
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using webrtc::MockStatsObserver;
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using webrtc::ObserverInterface;
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using webrtc::PeerConnection;
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using webrtc::PeerConnectionInterface;
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using ::testing::Values;
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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using webrtc::PeerConnectionFactory;
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using webrtc::PeerConnectionProxy;
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using webrtc::RTCErrorType;
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using webrtc::RTCTransportStats;
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using webrtc::RtpSenderInterface;
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using webrtc::RtpReceiverInterface;
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using webrtc::RtpSenderInterface;
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using webrtc::RtpTransceiverDirection;
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using webrtc::RtpTransceiverInit;
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using webrtc::RtpTransceiverInterface;
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using webrtc::SdpSemantics;
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using webrtc::SdpType;
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using webrtc::SessionDescriptionInterface;
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using webrtc::StreamCollectionInterface;
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using webrtc::VideoTrackInterface;
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namespace {
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static const int kDefaultTimeout = 10000;
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static const int kMaxWaitForStatsMs = 3000;
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@ -5090,5 +5055,6 @@ TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
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}
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} // namespace
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} // namespace webrtc
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#endif // if !defined(THREAD_SANITIZER)
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@ -94,6 +94,9 @@
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#include "pc/test/androidtestinitializer.h"
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#endif
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namespace webrtc {
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namespace {
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static const char kStreamId1[] = "local_stream_1";
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static const char kStreamId2[] = "local_stream_2";
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static const char kStreamId3[] = "local_stream_3";
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@ -449,49 +452,13 @@ static const char kDtlsSdesFallbackSdp[] =
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"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
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"dummy_session_params\r\n";
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using ::cricket::StreamParams;
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using ::testing::Exactly;
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using ::testing::Values;
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using cricket::StreamParams;
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using webrtc::AudioSourceInterface;
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using webrtc::AudioTrack;
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using webrtc::AudioTrackInterface;
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using webrtc::DataBuffer;
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using webrtc::DataChannelInterface;
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using webrtc::IceCandidateInterface;
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using webrtc::MediaStream;
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using webrtc::MediaStreamInterface;
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using webrtc::MediaStreamTrackInterface;
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using webrtc::MockCreateSessionDescriptionObserver;
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using webrtc::MockDataChannelObserver;
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using webrtc::MockPeerConnectionObserver;
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using webrtc::MockSetSessionDescriptionObserver;
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using webrtc::MockStatsObserver;
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using webrtc::NotifierInterface;
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using webrtc::ObserverInterface;
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using webrtc::PeerConnectionInterface;
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using webrtc::PeerConnectionObserver;
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using webrtc::RTCError;
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using webrtc::RTCErrorType;
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using webrtc::RtpReceiverInterface;
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using webrtc::RtpSenderInterface;
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using webrtc::RtpSenderProxyWithInternal;
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using webrtc::RtpSenderInternal;
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using webrtc::RtpTransceiverDirection;
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using webrtc::SdpParseError;
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using webrtc::SdpSemantics;
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using webrtc::SdpType;
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using webrtc::SessionDescriptionInterface;
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using webrtc::StreamCollection;
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using webrtc::StreamCollectionInterface;
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using webrtc::VideoTrackSourceInterface;
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using webrtc::VideoTrack;
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using webrtc::VideoTrackInterface;
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
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namespace {
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// Gets the first ssrc of given content type from the ContentInfo.
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bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
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if (!content_info || !ssrc) {
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@ -660,8 +627,6 @@ class MockTrackObserver : public ObserverInterface {
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NotifierInterface* notifier_;
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};
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} // namespace
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// The PeerConnectionMediaConfig tests below verify that configuration and
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// constraints are propagated into the PeerConnection's MediaConfig. These
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// settings are intended for MediaChannel constructors, but that is not
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@ -4104,3 +4069,6 @@ TEST(RTCConfigurationTest, ComparisonOperators) {
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PeerConnectionInterface::RTCConfigurationType::kAggressive);
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EXPECT_NE(a, h);
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}
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} // namespace
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} // namespace webrtc
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