Fully qualify googletest symbols.
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}diff --git a/pc/channel_manager_unittest.cc b/pc/channel_manager_unittest.cc
index 03c68ea..8a20f40 100644
--- a/pc/channel_manager_unittest.cc
+++ b/pc/channel_manager_unittest.cc
@@ -40,7 +40,7 @@
VideoCodec(99, "H264"), VideoCodec(100, "VP8"), VideoCodec(96, "rtx"),
};
-class ChannelManagerTest : public testing::Test {
+class ChannelManagerTest : public ::testing::Test {
protected:
ChannelManagerTest()
: network_(rtc::Thread::CreateWithSocketServer()),
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index edcea88..9c5f82b 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -101,7 +101,7 @@
// Base class for Voice/Video/RtpDataChannel tests
template <class T>
-class ChannelTest : public testing::Test, public sigslot::has_slots<> {
+class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
public:
enum Flags {
RTCP_MUX = 0x1,
diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
index 7ce40fb..52c54e7 100644
--- a/pc/data_channel_unittest.cc
+++ b/pc/data_channel_unittest.cc
@@ -62,7 +62,7 @@
// TODO(deadbeef): The fact that these tests use a fake provider makes them not
// too valuable. Should rewrite using the
// peerconnection_datachannel_unittest.cc infrastructure.
-class SctpDataChannelTest : public testing::Test {
+class SctpDataChannelTest : public ::testing::Test {
protected:
SctpDataChannelTest()
: provider_(new FakeDataChannelProvider()),
@@ -613,7 +613,7 @@
webrtc_data_channel_->state(), kDefaultTimeout);
}
-class SctpSidAllocatorTest : public testing::Test {
+class SctpSidAllocatorTest : public ::testing::Test {
protected:
SctpSidAllocator allocator_;
};
diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index abf1b2d..d4ad5fa 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -39,7 +39,7 @@
const int kRtpAuthTagLen = 10;
-class DtlsSrtpTransportTest : public testing::Test,
+class DtlsSrtpTransportTest : public ::testing::Test,
public sigslot::has_slots<> {
protected:
DtlsSrtpTransportTest() {}
diff --git a/pc/dtls_transport_unittest.cc b/pc/dtls_transport_unittest.cc
index 508e557..0c27c1b 100644
--- a/pc/dtls_transport_unittest.cc
+++ b/pc/dtls_transport_unittest.cc
@@ -47,7 +47,7 @@
std::vector<DtlsTransportState> states_;
};
-class DtlsTransportTest : public testing::Test {
+class DtlsTransportTest : public ::testing::Test {
public:
DtlsTransport* transport() { return transport_.get(); }
DtlsTransportObserverInterface* observer() { return &observer_; }
diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc
index b98b035..069833a 100644
--- a/pc/dtmf_sender_unittest.cc
+++ b/pc/dtmf_sender_unittest.cc
@@ -114,7 +114,7 @@
sigslot::signal0<> SignalDestroyed;
};
-class DtmfSenderTest : public testing::Test {
+class DtmfSenderTest : public ::testing::Test {
protected:
DtmfSenderTest()
: observer_(new rtc::RefCountedObject<FakeDtmfObserver>()),
diff --git a/pc/ice_server_parsing_unittest.cc b/pc/ice_server_parsing_unittest.cc
index 9f1b817..2262903 100644
--- a/pc/ice_server_parsing_unittest.cc
+++ b/pc/ice_server_parsing_unittest.cc
@@ -19,7 +19,7 @@
namespace webrtc {
-class IceServerParsingTest : public testing::Test {
+class IceServerParsingTest : public ::testing::Test {
public:
// Convenience functions for parsing a single URL. Result is stored in
// |stun_servers_| and |turn_servers_|.
diff --git a/pc/ice_transport_unittest.cc b/pc/ice_transport_unittest.cc
index 8d668de..62d7953 100644
--- a/pc/ice_transport_unittest.cc
+++ b/pc/ice_transport_unittest.cc
@@ -24,7 +24,7 @@
namespace webrtc {
-class IceTransportTest : public testing::Test {};
+class IceTransportTest : public ::testing::Test {};
TEST_F(IceTransportTest, CreateNonSelfDeletingTransport) {
auto cricket_transport =
diff --git a/pc/jsep_session_description_unittest.cc b/pc/jsep_session_description_unittest.cc
index 6ac1a00..674b6e7 100644
--- a/pc/jsep_session_description_unittest.cc
+++ b/pc/jsep_session_description_unittest.cc
@@ -82,7 +82,7 @@
return desc;
}
-class JsepSessionDescriptionTest : public testing::Test {
+class JsepSessionDescriptionTest : public ::testing::Test {
protected:
virtual void SetUp() {
int port = 1234;
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index 64da95b..b18f85a 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -78,7 +78,7 @@
};
class JsepTransportControllerTest : public JsepTransportController::Observer,
- public testing::Test,
+ public ::testing::Test,
public sigslot::has_slots<> {
public:
JsepTransportControllerTest() : signaling_thread_(rtc::Thread::Current()) {
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 9ed079b..ac17d6f 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -40,7 +40,7 @@
SdpType remote_type;
};
-class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
+class JsepTransport2Test : public ::testing::Test, public sigslot::has_slots<> {
protected:
std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport(
rtc::PacketTransportInternal* rtp_packet_transport,
@@ -156,7 +156,7 @@
// The parameterized tests cover both cases when RTCP mux is enable and
// disabled.
class JsepTransport2WithRtcpMux : public JsepTransport2Test,
- public testing::WithParamInterface<bool> {};
+ public ::testing::WithParamInterface<bool> {};
// This test verifies the ICE parameters are properly applied to the transports.
TEST_P(JsepTransport2WithRtcpMux, SetIceParameters) {
@@ -640,7 +640,7 @@
INSTANTIATE_TEST_SUITE_P(JsepTransport2Test,
JsepTransport2WithRtcpMux,
- testing::Bool());
+ ::testing::Bool());
// Test that a reoffer in the opposite direction is successful as long as the
// role isn't changing. Doesn't test every possible combination like the test
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index 7c0a7de..efd2318 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -82,17 +82,17 @@
using rtc::CS_AES_CM_128_HMAC_SHA1_32;
using rtc::CS_AES_CM_128_HMAC_SHA1_80;
using rtc::UniqueRandomIdGenerator;
-using testing::Contains;
-using testing::Each;
-using testing::ElementsAreArray;
-using testing::Eq;
-using testing::Field;
-using testing::IsEmpty;
-using testing::IsFalse;
-using testing::Ne;
-using testing::Not;
-using testing::Pointwise;
-using testing::SizeIs;
+using ::testing::Contains;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::IsEmpty;
+using ::testing::IsFalse;
+using ::testing::Ne;
+using ::testing::Not;
+using ::testing::Pointwise;
+using ::testing::SizeIs;
using webrtc::RtpExtension;
using webrtc::RtpTransceiverDirection;
@@ -405,7 +405,7 @@
// was designed for Plan B SDP, where only one audio "m=" section and one video
// "m=" section could be generated, and ordering couldn't be controlled. Many of
// these tests may be obsolete as a result, and should be refactored or removed.
-class MediaSessionDescriptionFactoryTest : public testing::Test {
+class MediaSessionDescriptionFactoryTest : public ::testing::Test {
public:
MediaSessionDescriptionFactoryTest()
: f1_(&tdf1_, &ssrc_generator1), f2_(&tdf2_, &ssrc_generator2) {
diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc
index 4b25e78..b49481e 100644
--- a/pc/media_stream_unittest.cc
+++ b/pc/media_stream_unittest.cc
@@ -50,7 +50,7 @@
NotifierInterface* notifier_;
};
-class MediaStreamTest : public testing::Test {
+class MediaStreamTest : public ::testing::Test {
protected:
virtual void SetUp() {
stream_ = MediaStream::Create(kStreamId1);
diff --git a/pc/peer_connection_end_to_end_unittest.cc b/pc/peer_connection_end_to_end_unittest.cc
index a303199..4cd59c8 100644
--- a/pc/peer_connection_end_to_end_unittest.cc
+++ b/pc/peer_connection_end_to_end_unittest.cc
@@ -33,11 +33,11 @@
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_audio_encoder_factory.h"
-using testing::AtLeast;
-using testing::Invoke;
-using testing::StrictMock;
-using testing::Values;
-using testing::_;
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::StrictMock;
+using ::testing::Values;
using webrtc::DataChannelInterface;
using webrtc::MediaStreamInterface;
@@ -51,7 +51,7 @@
} // namespace
class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
- public testing::Test {
+ public ::testing::Test {
public:
typedef std::vector<rtc::scoped_refptr<DataChannelInterface>> DataChannelList;
diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc
index 41a5408..a19e430 100644
--- a/pc/peer_connection_factory_unittest.cc
+++ b/pc/peer_connection_factory_unittest.cc
@@ -98,7 +98,7 @@
} // namespace
-class PeerConnectionFactoryTest : public testing::Test {
+class PeerConnectionFactoryTest : public ::testing::Test {
void SetUp() {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc
index 807a8e1..03c1128 100644
--- a/pc/peer_connection_ice_unittest.cc
+++ b/pc/peer_connection_ice_unittest.cc
@@ -975,7 +975,7 @@
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
-class PeerConnectionIceConfigTest : public testing::Test {
+class PeerConnectionIceConfigTest : public ::testing::Test {
protected:
void SetUp() override {
pc_factory_ = CreatePeerConnectionFactory(
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index 58c3439..8aa7675 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -1123,7 +1123,7 @@
// virtual network, fake A/V capture and fake encoder/decoders. The
// PeerConnections share the threads/socket servers, but use separate versions
// of everything else (including "PeerConnectionFactory"s).
-class PeerConnectionIntegrationBaseTest : public testing::Test {
+class PeerConnectionIntegrationBaseTest : public ::testing::Test {
public:
explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
: sdp_semantics_(sdp_semantics),
@@ -4841,8 +4841,8 @@
ConnectFakeSignaling();
auto output = absl::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
- ON_CALL(*output, IsActive()).WillByDefault(testing::Return(true));
- ON_CALL(*output, Write(::testing::_)).WillByDefault(testing::Return(true));
+ ON_CALL(*output, IsActive()).WillByDefault(::testing::Return(true));
+ ON_CALL(*output, Write(::testing::_)).WillByDefault(::testing::Return(true));
EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
std::move(output), webrtc::RtcEventLog::kImmediateOutput));
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index edede25..21f6a05 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -667,7 +667,7 @@
};
// TODO(steveanton): Convert to use the new PeerConnectionWrapper.
-class PeerConnectionInterfaceBaseTest : public testing::Test {
+class PeerConnectionInterfaceBaseTest : public ::testing::Test {
protected:
explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
@@ -3959,7 +3959,7 @@
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
-class PeerConnectionMediaConfigTest : public testing::Test {
+class PeerConnectionMediaConfigTest : public ::testing::Test {
protected:
void SetUp() override {
pcf_ = PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc
index 868ff67..26d702b 100644
--- a/pc/peer_connection_jsep_unittest.cc
+++ b/pc/peer_connection_jsep_unittest.cc
@@ -710,7 +710,7 @@
// - The new transceiver is associated with the new MID value.
class RecycleMediaSectionTest
: public PeerConnectionJsepTest,
- public testing::WithParamInterface<
+ public ::testing::WithParamInterface<
std::tuple<cricket::MediaType, cricket::MediaType>> {
protected:
RecycleMediaSectionTest() {
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index 48c14e3..c03bab4 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -87,7 +87,7 @@
MethodFunctor on_success_;
};
-class PeerConnectionRtpBaseTest : public testing::Test {
+class PeerConnectionRtpBaseTest : public ::testing::Test {
public:
explicit PeerConnectionRtpBaseTest(SdpSemantics sdp_semantics)
: sdp_semantics_(sdp_semantics),
diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc
index 3efc6cb..5eaa501 100644
--- a/pc/peer_connection_simulcast_unittest.cc
+++ b/pc/peer_connection_simulcast_unittest.cc
@@ -87,7 +87,7 @@
} // namespace
namespace webrtc {
-class PeerConnectionSimulcastTests : public testing::Test {
+class PeerConnectionSimulcastTests : public ::testing::Test {
public:
PeerConnectionSimulcastTests()
: pc_factory_(
diff --git a/pc/playout_latency_unittest.cc b/pc/playout_latency_unittest.cc
index c991254..db139f4 100644
--- a/pc/playout_latency_unittest.cc
+++ b/pc/playout_latency_unittest.cc
@@ -27,7 +27,7 @@
namespace webrtc {
-class PlayoutLatencyTest : public testing::Test {
+class PlayoutLatencyTest : public ::testing::Test {
public:
PlayoutLatencyTest()
: latency_(
diff --git a/pc/proxy_unittest.cc b/pc/proxy_unittest.cc
index f9b7063..a00b47f 100644
--- a/pc/proxy_unittest.cc
+++ b/pc/proxy_unittest.cc
@@ -87,7 +87,7 @@
END_PROXY_MAP()
#undef FakeProxy
-class SignalingProxyTest : public testing::Test {
+class SignalingProxyTest : public ::testing::Test {
public:
// Checks that the functions are called on the right thread.
void CheckSignalingThread() { EXPECT_TRUE(signaling_thread_->IsCurrent()); }
@@ -173,7 +173,7 @@
EXPECT_EQ("Method2", fake_signaling_proxy_->Method2(arg1, arg2));
}
-class ProxyTest : public testing::Test {
+class ProxyTest : public ::testing::Test {
public:
// Checks that the functions are called on the right thread.
void CheckSignalingThread() { EXPECT_TRUE(signaling_thread_->IsCurrent()); }
@@ -274,7 +274,7 @@
PROXY_METHOD0(void, Bar)
END_PROXY_MAP()
-class OwnedProxyTest : public testing::Test {
+class OwnedProxyTest : public ::testing::Test {
public:
OwnedProxyTest()
: signaling_thread_(rtc::Thread::Create()),
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 5b0d164..7e843a6 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -38,9 +38,9 @@
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
-using testing::AtLeast;
-using testing::Invoke;
-using testing::Return;
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::Return;
namespace webrtc {
@@ -496,7 +496,7 @@
rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
};
-class RTCStatsCollectorTest : public testing::Test {
+class RTCStatsCollectorTest : public ::testing::Test {
public:
RTCStatsCollectorTest()
: pc_(new rtc::RefCountedObject<FakePeerConnectionForStats>()),
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 174e78e..6935769 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -103,7 +103,7 @@
RTCStatsReportTraceListener* RTCStatsReportTraceListener::traced_report_ =
nullptr;
-class RTCStatsIntegrationTest : public testing::Test {
+class RTCStatsIntegrationTest : public ::testing::Test {
public:
RTCStatsIntegrationTest()
: network_thread_(new rtc::Thread(&virtual_socket_server_)),
diff --git a/pc/rtc_stats_traversal_unittest.cc b/pc/rtc_stats_traversal_unittest.cc
index c08d293..c7d0971 100644
--- a/pc/rtc_stats_traversal_unittest.cc
+++ b/pc/rtc_stats_traversal_unittest.cc
@@ -22,7 +22,7 @@
namespace webrtc {
-class RTCStatsTraversalTest : public testing::Test {
+class RTCStatsTraversalTest : public ::testing::Test {
public:
RTCStatsTraversalTest() {
transport_ = new RTCTransportStats("transport", 0);
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index e6d0083..38f414c 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -87,8 +87,8 @@
namespace webrtc {
class RtpSenderReceiverTest
- : public testing::Test,
- public testing::WithParamInterface<std::pair<RidList, RidList>>,
+ : public ::testing::Test,
+ public ::testing::WithParamInterface<std::pair<RidList, RidList>>,
public sigslot::has_slots<> {
public:
RtpSenderReceiverTest()
diff --git a/pc/sctp_transport_unittest.cc b/pc/sctp_transport_unittest.cc
index e2bdf44..3771b2c 100644
--- a/pc/sctp_transport_unittest.cc
+++ b/pc/sctp_transport_unittest.cc
@@ -76,7 +76,7 @@
std::vector<SctpTransportState> states_;
};
-class SctpTransportTest : public testing::Test {
+class SctpTransportTest : public ::testing::Test {
public:
SctpTransport* transport() { return transport_.get(); }
SctpTransportObserverInterface* observer() { return &observer_; }
diff --git a/pc/sctp_utils_unittest.cc b/pc/sctp_utils_unittest.cc
index 01f5434..70c6277 100644
--- a/pc/sctp_utils_unittest.cc
+++ b/pc/sctp_utils_unittest.cc
@@ -16,7 +16,7 @@
#include "rtc_base/copy_on_write_buffer.h"
#include "test/gtest.h"
-class SctpUtilsTest : public testing::Test {
+class SctpUtilsTest : public ::testing::Test {
public:
void VerifyOpenMessageFormat(const rtc::CopyOnWriteBuffer& packet,
const std::string& label,
diff --git a/pc/srtp_filter_unittest.cc b/pc/srtp_filter_unittest.cc
index d7368d1..7964002 100644
--- a/pc/srtp_filter_unittest.cc
+++ b/pc/srtp_filter_unittest.cc
@@ -62,7 +62,7 @@
kTestKeyParamsGcm4,
"");
-class SrtpFilterTest : public testing::Test {
+class SrtpFilterTest : public ::testing::Test {
protected:
SrtpFilterTest() {}
static std::vector<CryptoParams> MakeVector(const CryptoParams& params) {
diff --git a/pc/srtp_session_unittest.cc b/pc/srtp_session_unittest.cc
index 8ca5997..8feceb4 100644
--- a/pc/srtp_session_unittest.cc
+++ b/pc/srtp_session_unittest.cc
@@ -29,7 +29,7 @@
std::vector<int> kEncryptedHeaderExtensionIds;
-class SrtpSessionTest : public testing::Test {
+class SrtpSessionTest : public ::testing::Test {
public:
SrtpSessionTest() { webrtc::metrics::Reset(); }
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index f3fab89..a12d2c5 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -42,7 +42,7 @@
"rqponmlkjihgfedcbaZYXWVUTSRQPONMLKJIHGFEDCBA";
static const int kTestKeyGcm256Len = 44; // 256 bits key + 96 bits salt.
-class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
+class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
protected:
SrtpTransportTest() {
bool rtcp_mux_enabled = true;
@@ -336,7 +336,7 @@
class SrtpTransportTestWithExternalAuth
: public SrtpTransportTest,
- public testing::WithParamInterface<bool> {};
+ public ::testing::WithParamInterface<bool> {};
TEST_P(SrtpTransportTestWithExternalAuth,
SendAndRecvPacket_AES_CM_128_HMAC_SHA1_80) {
diff --git a/pc/stats_collector_unittest.cc b/pc/stats_collector_unittest.cc
index 7ebac7f..3118ef9 100644
--- a/pc/stats_collector_unittest.cc
+++ b/pc/stats_collector_unittest.cc
@@ -590,7 +590,7 @@
double time_now_;
};
-class StatsCollectorTest : public testing::Test {
+class StatsCollectorTest : public ::testing::Test {
protected:
rtc::scoped_refptr<FakePeerConnectionForStats> CreatePeerConnection() {
return new rtc::RefCountedObject<FakePeerConnectionForStats>();
diff --git a/pc/test/fake_audio_capture_module_unittest.cc b/pc/test/fake_audio_capture_module_unittest.cc
index 0130af1..c8be683 100644
--- a/pc/test/fake_audio_capture_module_unittest.cc
+++ b/pc/test/fake_audio_capture_module_unittest.cc
@@ -18,7 +18,7 @@
#include "rtc_base/gunit.h"
#include "test/gtest.h"
-class FakeAdmTest : public testing::Test, public webrtc::AudioTransport {
+class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
protected:
static const int kMsInSecond = 1000;
diff --git a/pc/test/mock_data_channel.h b/pc/test/mock_data_channel.h
index 2db0c36..3385ec2 100644
--- a/pc/test/mock_data_channel.h
+++ b/pc/test/mock_data_channel.h
@@ -31,17 +31,17 @@
uint32_t messages_received,
uint64_t bytes_received)
: rtc::RefCountedObject<DataChannel>(nullptr, cricket::DCT_NONE, label) {
- EXPECT_CALL(*this, id()).WillRepeatedly(testing::Return(id));
- EXPECT_CALL(*this, state()).WillRepeatedly(testing::Return(state));
- EXPECT_CALL(*this, protocol()).WillRepeatedly(testing::Return(protocol));
+ EXPECT_CALL(*this, id()).WillRepeatedly(::testing::Return(id));
+ EXPECT_CALL(*this, state()).WillRepeatedly(::testing::Return(state));
+ EXPECT_CALL(*this, protocol()).WillRepeatedly(::testing::Return(protocol));
EXPECT_CALL(*this, messages_sent())
- .WillRepeatedly(testing::Return(messages_sent));
+ .WillRepeatedly(::testing::Return(messages_sent));
EXPECT_CALL(*this, bytes_sent())
- .WillRepeatedly(testing::Return(bytes_sent));
+ .WillRepeatedly(::testing::Return(bytes_sent));
EXPECT_CALL(*this, messages_received())
- .WillRepeatedly(testing::Return(messages_received));
+ .WillRepeatedly(::testing::Return(messages_received));
EXPECT_CALL(*this, bytes_received())
- .WillRepeatedly(testing::Return(bytes_received));
+ .WillRepeatedly(::testing::Return(bytes_received));
}
MOCK_CONST_METHOD0(id, int());
MOCK_CONST_METHOD0(state, DataState());
diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc
index 9edd60e..3b0022c 100644
--- a/pc/track_media_info_map_unittest.cc
+++ b/pc/track_media_info_map_unittest.cc
@@ -55,13 +55,13 @@
rtc::scoped_refptr<MockRtpSenderInternal> sender(
new rtc::RefCountedObject<MockRtpSenderInternal>());
EXPECT_CALL(*sender, track())
- .WillRepeatedly(testing::Return(std::move(track)));
- EXPECT_CALL(*sender, ssrc()).WillRepeatedly(testing::Return(first_ssrc));
+ .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));
+ .WillRepeatedly(::testing::Return(media_type));
EXPECT_CALL(*sender, GetParameters())
- .WillRepeatedly(testing::Return(CreateRtpParametersWithSsrcs(ssrcs)));
- EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(testing::Return(1));
+ .WillRepeatedly(::testing::Return(CreateRtpParametersWithSsrcs(ssrcs)));
+ EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(::testing::Return(1));
return sender;
}
@@ -72,16 +72,16 @@
rtc::scoped_refptr<MockRtpReceiverInternal> receiver(
new rtc::RefCountedObject<MockRtpReceiverInternal>());
EXPECT_CALL(*receiver, track())
- .WillRepeatedly(testing::Return(std::move(track)));
+ .WillRepeatedly(::testing::Return(std::move(track)));
EXPECT_CALL(*receiver, media_type())
- .WillRepeatedly(testing::Return(media_type));
+ .WillRepeatedly(::testing::Return(media_type));
EXPECT_CALL(*receiver, GetParameters())
- .WillRepeatedly(testing::Return(CreateRtpParametersWithSsrcs(ssrcs)));
- EXPECT_CALL(*receiver, AttachmentId()).WillRepeatedly(testing::Return(1));
+ .WillRepeatedly(::testing::Return(CreateRtpParametersWithSsrcs(ssrcs)));
+ EXPECT_CALL(*receiver, AttachmentId()).WillRepeatedly(::testing::Return(1));
return receiver;
}
-class TrackMediaInfoMapTest : public testing::Test {
+class TrackMediaInfoMapTest : public ::testing::Test {
public:
TrackMediaInfoMapTest()
: voice_media_info_(new cricket::VoiceMediaInfo()),
diff --git a/pc/video_track_unittest.cc b/pc/video_track_unittest.cc
index c9f9bcb..fc05f66 100644
--- a/pc/video_track_unittest.cc
+++ b/pc/video_track_unittest.cc
@@ -27,7 +27,7 @@
using webrtc::VideoTrackInterface;
using webrtc::VideoTrackSource;
-class VideoTrackTest : public testing::Test {
+class VideoTrackTest : public ::testing::Test {
public:
VideoTrackTest() : frame_source_(640, 480, rtc::kNumMicrosecsPerSec / 30) {
static const char kVideoTrackId[] = "track_id";
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 6b38689..a700a5a 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -998,7 +998,7 @@
// WebRtcSdpTest
-class WebRtcSdpTest : public testing::Test {
+class WebRtcSdpTest : public ::testing::Test {
public:
WebRtcSdpTest() : jdesc_(kDummyType) {
#ifdef WEBRTC_ANDROID
@@ -2989,7 +2989,7 @@
}
class WebRtcSdpExtmapTest : public WebRtcSdpTest,
- public testing::WithParamInterface<bool> {};
+ public ::testing::WithParamInterface<bool> {};
TEST_P(WebRtcSdpExtmapTest,
DeserializeSessionDescriptionWithSessionLevelExtmap) {
