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 <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26831}diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index bc8a320..86c0b76 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -44,7 +44,7 @@
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 60bef79..c86e906 100644
--- a/common_video/h264/sps_vui_rewriter_unittest.cc
+++ b/common_video/h264/sps_vui_rewriter_unittest.cc
@@ -171,14 +171,14 @@
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 58286b0..3c06857 100644
--- a/examples/peerconnection/client/conductor.h
+++ b/examples/peerconnection/client/conductor.h
@@ -63,7 +63,7 @@
//
void OnSignalingChange(
- webrtc::PeerConnectionInterface::SignalingState new_state) override{};
+ webrtc::PeerConnectionInterface::SignalingState new_state) override {}
void OnAddTrack(
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
@@ -74,9 +74,9 @@
rtc::scoped_refptr<webrtc::DataChannelInterface> 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 a609f98..edaf798 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 @@
"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 bf757f6..1a629a8 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 @@
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 @@
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 f618c0d..82a6a64 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 @@
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 d9c22d7..cdfc706 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -65,7 +65,7 @@
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 292193d..ecf2963 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 @@
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 69e6e50..d5202fb 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 @@
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 3f1e059..267213e 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -104,7 +104,7 @@
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 @@
void GetMetrics(EchoControl::Metrics* metrics) const override {}
- void SetAudioBufferDelay(size_t delay_ms) override{};
+ void SetAudioBufferDelay(size_t delay_ms) override {}
private:
std::deque<std::vector<std::vector<float>>> received_render_blocks_;
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index abceeec..69fc779 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -133,7 +133,7 @@
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 0be0c75..5324d5e 100644
--- a/modules/video_coding/generic_encoder_unittest.cc
+++ b/modules/video_coding/generic_encoder_unittest.cc
@@ -42,7 +42,7 @@
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 f628f82..51aeb40 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -214,8 +214,8 @@
}
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
- const rtc::VideoSinkWants& wants) override{};
- void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override{};
+ const rtc::VideoSinkWants& wants) override {}
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
VideoTrackSourceInterface* GetSource() const override { return nullptr; }
};
@@ -2197,7 +2197,7 @@
RTCStatsMember<int32_t> 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 82fa435..700cb1f 100644
--- a/rtc_base/ssl_stream_adapter_unittest.cc
+++ b/rtc_base/ssl_stream_adapter_unittest.cc
@@ -731,7 +731,7 @@
break;
}
}
- };
+ }
void ReadData(rtc::StreamInterface* stream) override {
char buffer[1600];
@@ -880,7 +880,7 @@
RTC_LOG(LS_INFO) << "Sent " << sent_ << " packets; received "
<< received_.size();
}
- };
+ }
private:
BufferQueueStream client_buffer_;
@@ -907,7 +907,7 @@
}
return test_base_->DataWritten(this, data, data_len, written, error);
-};
+}
class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS {
public:
@@ -919,7 +919,7 @@
// certificate.
class SSLStreamAdapterTestDTLSCertChain : public SSLStreamAdapterTestDTLS {
public:
- SSLStreamAdapterTestDTLSCertChain() : SSLStreamAdapterTestDTLS("", ""){};
+ SSLStreamAdapterTestDTLSCertChain() : SSLStreamAdapterTestDTLS("", "") {}
void SetUp() override {
CreateStreams();
@@ -950,7 +950,7 @@
// Test that we can make a handshake work
TEST_P(SSLStreamAdapterTestTLS, TestTLSConnect) {
TestHandshake();
-};
+}
TEST_P(SSLStreamAdapterTestTLS, GetPeerCertChainWithOneCertificate) {
TestHandshake();
@@ -1009,13 +1009,13 @@
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 @@
// 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 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, TestDTLSConnectWithLostFirstPacket) {
SetLoseFirstPacket(true);
TestHandshake();
-};
+}
// Test a handshake with loss and delay
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) {
@@ -1109,7 +1109,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
&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 a7d4a42..2081364 100644
--- a/stats/rtc_stats_report_unittest.cc
+++ b/stats/rtc_stats_report_unittest.cc
@@ -26,7 +26,7 @@
RTCStatsMember<int32_t> 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 @@
RTCStatsMember<double> 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 @@
RTCStatsMember<std::string> 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<RTCStatsReport> report = RTCStatsReport::Create(1337);
diff --git a/stats/rtc_stats_unittest.cc b/stats/rtc_stats_unittest.cc
index b079ddd..0755660 100644
--- a/stats/rtc_stats_unittest.cc
+++ b/stats/rtc_stats_unittest.cc
@@ -49,7 +49,7 @@
RTCStatsMember<int32_t> 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 @@
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 b2674a8..14bc6e0 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 @@
}
}
-}; // namespace webrtc
+} // namespace webrtc
diff --git a/test/scenario/network/traffic_route.cc b/test/scenario/network/traffic_route.cc
index 67a2cb3..d82e292 100644
--- a/test/scenario/network/traffic_route.cc
+++ b/test/scenario/network/traffic_route.cc
@@ -23,7 +23,7 @@
class NullReceiver : public EmulatedNetworkReceiverInterface {
public:
- void OnPacketReceived(EmulatedIpPacket packet) override{};
+ void OnPacketReceived(EmulatedIpPacket packet) override {}
};
class ActionReceiver : public EmulatedNetworkReceiverInterface {
@@ -36,7 +36,7 @@
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.