From c4dd730765fccd0ac10c7ad3bc6dbedc8848699b Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Mon, 25 Feb 2019 09:12:02 +0100 Subject: [PATCH] Fix -Wextra-semi warnings. Starting from https://chromium-review.googlesource.com/c/1485012, -Wextra-semi is enabled and WebRTC has some violations to fix. This is a follow-up of https://webrtc-review.googlesource.com/c/123560. Bug: webrtc:10355 Change-Id: I012b7497fc8991037fd77aa98f1579c22e08206f Reviewed-on: https://webrtc-review.googlesource.com/c/124126 Reviewed-by: Karl Wiberg Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#26831} --- api/test/fake_media_transport.h | 2 +- .../h264/sps_vui_rewriter_unittest.cc | 8 +-- examples/peerconnection/client/conductor.h | 6 +- .../acm2/audio_coding_module_unittest.cc | 2 +- .../codecs/opus/opus_speed_test.cc | 32 +++++----- .../neteq/tools/neteq_quality_test.h | 2 +- modules/audio_coding/test/EncodeDecodeTest.h | 2 +- .../linux/audio_device_alsa_linux.cc | 2 +- .../linux/audio_device_alsa_linux.h | 4 +- .../aec3/echo_canceller3_unittest.cc | 4 +- .../audio_processing_impl_unittest.cc | 2 +- .../video_coding/generic_encoder_unittest.cc | 2 +- pc/rtc_stats_collector_unittest.cc | 6 +- rtc_base/ssl_stream_adapter_unittest.cc | 58 +++++++++---------- stats/rtc_stats_report_unittest.cc | 6 +- stats/rtc_stats_unittest.cc | 4 +- .../source/rtp_to_ntp_estimator_unittest.cc | 2 +- test/scenario/network/traffic_route.cc | 4 +- 18 files changed, 74 insertions(+), 74 deletions(-) diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h index bc8a320f3d..86c0b76b59 100644 --- a/api/test/fake_media_transport.h +++ b/api/test/fake_media_transport.h @@ -44,7 +44,7 @@ class FakeMediaTransport : public MediaTransportInterface { RTCError RequestKeyFrame(uint64_t channel_id) override { return RTCError::OK(); - }; + } void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {} void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {} diff --git a/common_video/h264/sps_vui_rewriter_unittest.cc b/common_video/h264/sps_vui_rewriter_unittest.cc index 60bef79dee..c86e906e4d 100644 --- a/common_video/h264/sps_vui_rewriter_unittest.cc +++ b/common_video/h264/sps_vui_rewriter_unittest.cc @@ -171,14 +171,14 @@ void TestSps(SpsMode mode, SpsVuiRewriter::ParseResult expected_parse_result) { REWRITE_TEST(VuiAlreadyOptimal, kNoRewriteRequired_VuiOptimal, - SpsVuiRewriter::ParseResult::kVuiOk); + SpsVuiRewriter::ParseResult::kVuiOk) REWRITE_TEST(RewriteFullVui, kRewriteRequired_NoVui, - SpsVuiRewriter::ParseResult::kVuiRewritten); + SpsVuiRewriter::ParseResult::kVuiRewritten) REWRITE_TEST(AddBitstreamRestriction, kRewriteRequired_NoBitstreamRestriction, - SpsVuiRewriter::ParseResult::kVuiRewritten); + SpsVuiRewriter::ParseResult::kVuiRewritten) REWRITE_TEST(RewriteSuboptimalVui, kRewriteRequired_VuiSuboptimal, - SpsVuiRewriter::ParseResult::kVuiRewritten); + SpsVuiRewriter::ParseResult::kVuiRewritten) } // namespace webrtc diff --git a/examples/peerconnection/client/conductor.h b/examples/peerconnection/client/conductor.h index 58286b0a4b..3c06857a05 100644 --- a/examples/peerconnection/client/conductor.h +++ b/examples/peerconnection/client/conductor.h @@ -63,7 +63,7 @@ class Conductor : public webrtc::PeerConnectionObserver, // void OnSignalingChange( - webrtc::PeerConnectionInterface::SignalingState new_state) override{}; + webrtc::PeerConnectionInterface::SignalingState new_state) override {} void OnAddTrack( rtc::scoped_refptr receiver, const std::vector>& @@ -74,9 +74,9 @@ class Conductor : public webrtc::PeerConnectionObserver, rtc::scoped_refptr channel) override {} void OnRenegotiationNeeded() override {} void OnIceConnectionChange( - webrtc::PeerConnectionInterface::IceConnectionState new_state) override{}; + webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} void OnIceGatheringChange( - webrtc::PeerConnectionInterface::IceGatheringState new_state) override{}; + webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; void OnIceConnectionReceivingChange(bool receiving) override {} diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index a609f980e6..edaf798f2a 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -1484,7 +1484,7 @@ const std::string payload_checksum = "ab88b1a049c36bdfeb7e8b057ef6982a", "27fef7b799393347ec3b5694369a1c36", "27fef7b799393347ec3b5694369a1c36"); -}; // namespace +} // namespace TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc index bf757f6433..1a629a8c7c 100644 --- a/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -96,17 +96,17 @@ constexpr size_t kDurationSec = 400; EncodeDecode(kDurationSec); \ } -ADD_TEST(10); -ADD_TEST(9); -ADD_TEST(8); -ADD_TEST(7); -ADD_TEST(6); -ADD_TEST(5); -ADD_TEST(4); -ADD_TEST(3); -ADD_TEST(2); -ADD_TEST(1); -ADD_TEST(0); +ADD_TEST(10) +ADD_TEST(9) +ADD_TEST(8) +ADD_TEST(7) +ADD_TEST(6) +ADD_TEST(5) +ADD_TEST(4) +ADD_TEST(3) +ADD_TEST(2) +ADD_TEST(1) +ADD_TEST(0) #define ADD_BANDWIDTH_TEST(bandwidth) \ TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) { \ @@ -116,11 +116,11 @@ ADD_TEST(0); EncodeDecode(kDurationSec); \ } -ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND); -ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND); -ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND); -ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND); -ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND); +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND) // List all test cases: (channel, bit rat, filename, extension). const coding_param param_set[] = { diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h index f618c0dfec..82a6a64d00 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.h +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h @@ -34,7 +34,7 @@ enum LossModes { class LossModel { public: - virtual ~LossModel(){}; + virtual ~LossModel() {} virtual bool Lost(int now_ms) = 0; }; diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h index d9c22d7817..cdfc706eeb 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.h +++ b/modules/audio_coding/test/EncodeDecodeTest.h @@ -65,7 +65,7 @@ class Sender { class Receiver { public: Receiver(); - virtual ~Receiver() {}; + virtual ~Receiver() {} void Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, size_t channels, int file_num); void Teardown(); diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc index 292193d957..ecf296398a 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.cc +++ b/modules/audio_device/linux/audio_device_alsa_linux.cc @@ -50,7 +50,7 @@ void WebrtcAlsaErrorHandler(const char* file, const char* function, int err, const char* fmt, - ...){}; + ...) {} namespace webrtc { static const unsigned int ALSA_PLAYOUT_FREQ = 48000; diff --git a/modules/audio_device/linux/audio_device_alsa_linux.h b/modules/audio_device/linux/audio_device_alsa_linux.h index 69e6e50822..d5202fb166 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.h +++ b/modules/audio_device/linux/audio_device_alsa_linux.h @@ -131,8 +131,8 @@ class AudioDeviceLinuxALSA : public AudioDeviceGeneric { bool KeyPressed() const; - void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); }; - void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); }; + void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); } + void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); } inline int32_t InputSanityCheckAfterUnlockedPeriod() const; inline int32_t OutputSanityCheckAfterUnlockedPeriod() const; diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc index 3f1e059a0c..267213ea34 100644 --- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc +++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc @@ -104,7 +104,7 @@ class CaptureTransportVerificationProcessor : public BlockProcessor { void GetMetrics(EchoControl::Metrics* metrics) const override {} - void SetAudioBufferDelay(size_t delay_ms) override{}; + void SetAudioBufferDelay(size_t delay_ms) override {} private: RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CaptureTransportVerificationProcessor); @@ -134,7 +134,7 @@ class RenderTransportVerificationProcessor : public BlockProcessor { void GetMetrics(EchoControl::Metrics* metrics) const override {} - void SetAudioBufferDelay(size_t delay_ms) override{}; + void SetAudioBufferDelay(size_t delay_ms) override {} private: std::deque>> received_render_blocks_; diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc index abceeec250..69fc7793b4 100644 --- a/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/modules/audio_processing/audio_processing_impl_unittest.cc @@ -133,7 +133,7 @@ class TestRenderPreProcessor : public CustomProcessing { std::transform(channel_view.begin(), channel_view.end(), channel_view.begin(), ProcessSample); } - }; + } std::string ToString() const override { return "TestRenderPreProcessor"; } void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {} // Modifies a sample. This member is used in Process() to modify a frame and diff --git a/modules/video_coding/generic_encoder_unittest.cc b/modules/video_coding/generic_encoder_unittest.cc index 0be0c75913..5324d5ea78 100644 --- a/modules/video_coding/generic_encoder_unittest.cc +++ b/modules/video_coding/generic_encoder_unittest.cc @@ -42,7 +42,7 @@ class FakeEncodedImageCallback : public EncodedImageCallback { encoded_image.timing_.flags != VideoSendTiming::kNotTriggered; last_capture_timestamp_ = encoded_image.capture_time_ms_; return Result(Result::OK); - }; + } void OnDroppedFrame(DropReason reason) override { ++num_frames_dropped_; } diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc index f628f82454..51aeb406e2 100644 --- a/pc/rtc_stats_collector_unittest.cc +++ b/pc/rtc_stats_collector_unittest.cc @@ -214,8 +214,8 @@ class FakeVideoTrackForStats : public MediaStreamTrack { } void AddOrUpdateSink(rtc::VideoSinkInterface* sink, - const rtc::VideoSinkWants& wants) override{}; - void RemoveSink(rtc::VideoSinkInterface* sink) override{}; + const rtc::VideoSinkWants& wants) override {} + void RemoveSink(rtc::VideoSinkInterface* sink) override {} VideoTrackSourceInterface* GetSource() const override { return nullptr; } }; @@ -2197,7 +2197,7 @@ class RTCTestStats : public RTCStats { RTCStatsMember dummy_stat; }; -WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats", &dummy_stat); +WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats", &dummy_stat) // Overrides the stats collection to verify thread usage and that the resulting // partial reports are merged. diff --git a/rtc_base/ssl_stream_adapter_unittest.cc b/rtc_base/ssl_stream_adapter_unittest.cc index 82fa435908..700cb1f009 100644 --- a/rtc_base/ssl_stream_adapter_unittest.cc +++ b/rtc_base/ssl_stream_adapter_unittest.cc @@ -731,7 +731,7 @@ class SSLStreamAdapterTestTLS break; } } - }; + } void ReadData(rtc::StreamInterface* stream) override { char buffer[1600]; @@ -880,7 +880,7 @@ class SSLStreamAdapterTestDTLS RTC_LOG(LS_INFO) << "Sent " << sent_ << " packets; received " << received_.size(); } - }; + } private: BufferQueueStream client_buffer_; @@ -907,7 +907,7 @@ rtc::StreamResult SSLDummyStreamBase::Write(const void* data, } return test_base_->DataWritten(this, data, data_len, written, error); -}; +} class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS { public: @@ -919,7 +919,7 @@ class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS { // certificate. class SSLStreamAdapterTestDTLSCertChain : public SSLStreamAdapterTestDTLS { public: - SSLStreamAdapterTestDTLSCertChain() : SSLStreamAdapterTestDTLS("", ""){}; + SSLStreamAdapterTestDTLSCertChain() : SSLStreamAdapterTestDTLS("", "") {} void SetUp() override { CreateStreams(); @@ -950,7 +950,7 @@ class SSLStreamAdapterTestDTLSCertChain : public SSLStreamAdapterTestDTLS { // Test that we can make a handshake work TEST_P(SSLStreamAdapterTestTLS, TestTLSConnect) { TestHandshake(); -}; +} TEST_P(SSLStreamAdapterTestTLS, GetPeerCertChainWithOneCertificate) { TestHandshake(); @@ -1009,13 +1009,13 @@ TEST_P(SSLStreamAdapterTestTLS, TestTLSClose) { TestHandshake(); client_ssl_->Close(); EXPECT_EQ_WAIT(rtc::SS_CLOSED, server_ssl_->GetState(), handshake_wait_); -}; +} // Test transfer -- trivial TEST_P(SSLStreamAdapterTestTLS, TestTLSTransfer) { TestHandshake(); TestTransfer(100000); -}; +} // Test read-write after close. TEST_P(SSLStreamAdapterTestTLS, ReadWriteAfterClose) { @@ -1034,21 +1034,21 @@ TEST_P(SSLStreamAdapterTestTLS, ReadWriteAfterClose) { // But after closed read gives you EOS. rv = client_ssl_->Read(block, sizeof(block), &dummy, nullptr); ASSERT_EQ(rtc::SR_EOS, rv); -}; +} // Test a handshake with a bogus peer digest TEST_P(SSLStreamAdapterTestTLS, TestTLSBogusDigest) { SetPeerIdentitiesByDigest(false, true); TestHandshake(false); -}; +} TEST_P(SSLStreamAdapterTestTLS, TestTLSDelayedIdentity) { TestHandshakeWithDelayedIdentity(true); -}; +} TEST_P(SSLStreamAdapterTestTLS, TestTLSDelayedIdentityWithBogusDigest) { TestHandshakeWithDelayedIdentity(false); -}; +} // Test that the correct error is returned when SetPeerCertificateDigest is // called with an unknown algorithm. @@ -1093,7 +1093,7 @@ TEST_P(SSLStreamAdapterTestTLS, TestSetPeerCertificateDigestWithInvalidLength) { // Test that we can make a handshake work TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) { TestHandshake(); -}; +} // Test that we can make a handshake work if the first packet in // each direction is lost. This gives us predictable loss @@ -1101,7 +1101,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) { TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) { SetLoseFirstPacket(true); TestHandshake(); -}; +} // Test a handshake with loss and delay TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) { @@ -1109,7 +1109,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) { SetDelay(2000); SetHandshakeWait(20000); TestHandshake(); -}; +} // Test a handshake with small MTU // Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3910 @@ -1117,34 +1117,34 @@ TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) { SetMtu(700); SetHandshakeWait(20000); TestHandshake(); -}; +} // Test transfer -- trivial TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) { TestHandshake(); TestTransfer(100); -}; +} TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) { TestHandshake(); SetLoss(10); TestTransfer(100); -}; +} TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) { SetDamage(); // Must be called first because first packet // write happens at end of handshake. TestHandshake(); TestTransfer(100); -}; +} TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentity) { TestHandshakeWithDelayedIdentity(true); -}; +} TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentityWithBogusDigest) { TestHandshakeWithDelayedIdentity(false); -}; +} // Test DTLS-SRTP with all high ciphers TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) { @@ -1161,7 +1161,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80); -}; +} // Test DTLS-SRTP with all low ciphers TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) { @@ -1178,7 +1178,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_32); -}; +} // Test DTLS-SRTP with a mismatch -- should not converge TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) { @@ -1194,7 +1194,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) { ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); int server_cipher; ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); -}; +} // Test DTLS-SRTP with each side being mixed -- should select high TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) { @@ -1212,7 +1212,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80); -}; +} // Test DTLS-SRTP with all GCM-128 ciphers. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) { @@ -1229,7 +1229,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AEAD_AES_128_GCM); -}; +} // Test DTLS-SRTP with all GCM-256 ciphers. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) { @@ -1246,7 +1246,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AEAD_AES_256_GCM); -}; +} // Test DTLS-SRTP with mixed GCM-128/-256 ciphers -- should not converge. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) { @@ -1262,7 +1262,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) { ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher)); int server_cipher; ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher)); -}; +} // Test DTLS-SRTP with both GCM-128/-256 ciphers -- should select GCM-256. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMixed) { @@ -1280,7 +1280,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMixed) { ASSERT_EQ(client_cipher, server_cipher); ASSERT_EQ(client_cipher, rtc::SRTP_AEAD_AES_256_GCM); -}; +} // Test SRTP cipher suite lengths. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) { @@ -1309,7 +1309,7 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) { &key_len, &salt_len)); ASSERT_EQ(256 / 8, key_len); ASSERT_EQ(96 / 8, salt_len); -}; +} // Test an exporter TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) { diff --git a/stats/rtc_stats_report_unittest.cc b/stats/rtc_stats_report_unittest.cc index a7d4a42d49..2081364f66 100644 --- a/stats/rtc_stats_report_unittest.cc +++ b/stats/rtc_stats_report_unittest.cc @@ -26,7 +26,7 @@ class RTCTestStats1 : public RTCStats { RTCStatsMember integer; }; -WEBRTC_RTCSTATS_IMPL(RTCTestStats1, RTCStats, "test-stats-1", &integer); +WEBRTC_RTCSTATS_IMPL(RTCTestStats1, RTCStats, "test-stats-1", &integer) class RTCTestStats2 : public RTCStats { public: @@ -38,7 +38,7 @@ class RTCTestStats2 : public RTCStats { RTCStatsMember number; }; -WEBRTC_RTCSTATS_IMPL(RTCTestStats2, RTCStats, "test-stats-2", &number); +WEBRTC_RTCSTATS_IMPL(RTCTestStats2, RTCStats, "test-stats-2", &number) class RTCTestStats3 : public RTCStats { public: @@ -50,7 +50,7 @@ class RTCTestStats3 : public RTCStats { RTCStatsMember string; }; -WEBRTC_RTCSTATS_IMPL(RTCTestStats3, RTCStats, "test-stats-3", &string); +WEBRTC_RTCSTATS_IMPL(RTCTestStats3, RTCStats, "test-stats-3", &string) TEST(RTCStatsReport, AddAndGetStats) { rtc::scoped_refptr report = RTCStatsReport::Create(1337); diff --git a/stats/rtc_stats_unittest.cc b/stats/rtc_stats_unittest.cc index b079dddc74..0755660a0f 100644 --- a/stats/rtc_stats_unittest.cc +++ b/stats/rtc_stats_unittest.cc @@ -49,7 +49,7 @@ class RTCChildStats : public RTCStats { RTCStatsMember child_int; }; -WEBRTC_RTCSTATS_IMPL(RTCChildStats, RTCStats, "child-stats", &child_int); +WEBRTC_RTCSTATS_IMPL(RTCChildStats, RTCStats, "child-stats", &child_int) class RTCGrandChildStats : public RTCChildStats { public: @@ -64,7 +64,7 @@ class RTCGrandChildStats : public RTCChildStats { WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats, RTCChildStats, "grandchild-stats", - &grandchild_int); + &grandchild_int) TEST(RTCStatsTest, RTCStatsAndMembers) { RTCTestStats stats("testId", 42); diff --git a/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc b/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc index b2674a8346..14bc6e0a89 100644 --- a/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc +++ b/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc @@ -346,4 +346,4 @@ TEST(RtpToNtpTests, AveragesErrorOut) { } } -}; // namespace webrtc +} // namespace webrtc diff --git a/test/scenario/network/traffic_route.cc b/test/scenario/network/traffic_route.cc index 67a2cb349d..d82e2926eb 100644 --- a/test/scenario/network/traffic_route.cc +++ b/test/scenario/network/traffic_route.cc @@ -23,7 +23,7 @@ namespace { class NullReceiver : public EmulatedNetworkReceiverInterface { public: - void OnPacketReceived(EmulatedIpPacket packet) override{}; + void OnPacketReceived(EmulatedIpPacket packet) override {} }; class ActionReceiver : public EmulatedNetworkReceiverInterface { @@ -36,7 +36,7 @@ class ActionReceiver : public EmulatedNetworkReceiverInterface { RTC_DCHECK(port_); action_(); endpoint_->UnbindReceiver(port_.value()); - }; + } // We can't set port in constructor, because port will be provided by // endpoint, when this receiver will be binded to that endpoint.