webrtc: Remove semicolons.

Bug: chromium:926235
Change-Id: I66c10ab3df38adf87152d1f18cc8162afedca7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/123560
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26780}
diff --git a/api/peer_connection_factory_proxy.h b/api/peer_connection_factory_proxy.h
index a4527e9..620f1ca 100644
--- a/api/peer_connection_factory_proxy.h
+++ b/api/peer_connection_factory_proxy.h
@@ -31,17 +31,17 @@
               const PeerConnectionInterface::RTCConfiguration&,
               std::unique_ptr<cricket::PortAllocator>,
               std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
-              PeerConnectionObserver*);
+              PeerConnectionObserver*)
 PROXY_METHOD2(rtc::scoped_refptr<PeerConnectionInterface>,
               CreatePeerConnection,
               const PeerConnectionInterface::RTCConfiguration&,
-              PeerConnectionDependencies);
+              PeerConnectionDependencies)
 PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
                    GetRtpSenderCapabilities,
-                   cricket::MediaType);
+                   cricket::MediaType)
 PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
                    GetRtpReceiverCapabilities,
-                   cricket::MediaType);
+                   cricket::MediaType)
 PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
               CreateLocalMediaStream,
               const std::string&)
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index bc32566..7d9870b 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -160,15 +160,15 @@
                    streams)
 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
-PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
 PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
-PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
-PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
+PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
+PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
 PROXY_METHOD1(void,
               SetFrameDecryptor,
-              rtc::scoped_refptr<FrameDecryptorInterface>);
+              rtc::scoped_refptr<FrameDecryptorInterface>)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
-                   GetFrameDecryptor);
+                   GetFrameDecryptor)
 END_PROXY_MAP()
 
 }  // namespace webrtc
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 83cbe7a..f8a9757 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -106,14 +106,14 @@
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
 PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
-PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
 PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
-PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender)
 PROXY_METHOD1(void,
               SetFrameEncryptor,
-              rtc::scoped_refptr<FrameEncryptorInterface>);
+              rtc::scoped_refptr<FrameEncryptorInterface>)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
-                   GetFrameEncryptor);
+                   GetFrameEncryptor)
 END_PROXY_MAP()
 
 }  // namespace webrtc
diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h
index 049b74f..2407ce0 100644
--- a/api/stats/rtc_stats.h
+++ b/api/stats/rtc_stats.h
@@ -140,18 +140,16 @@
 //   }
 //
 #define WEBRTC_RTCSTATS_DECL()                                          \
- public:                                                                \
-  static const char kType[];                                            \
-                                                                        \
-  std::unique_ptr<webrtc::RTCStats> copy() const override;              \
-  const char* type() const override;                                    \
-                                                                        \
  protected:                                                             \
   std::vector<const webrtc::RTCStatsMemberInterface*>                   \
   MembersOfThisObjectAndAncestors(size_t local_var_additional_capacity) \
       const override;                                                   \
                                                                         \
- public:
+ public:                                                                \
+  static const char kType[];                                            \
+                                                                        \
+  std::unique_ptr<webrtc::RTCStats> copy() const override;              \
+  const char* type() const override
 
 #define WEBRTC_RTCSTATS_IMPL(this_class, parent_class, type_str, ...)          \
   const char this_class::kType[] = type_str;                                   \
diff --git a/common_audio/audio_converter.cc b/common_audio/audio_converter.cc
index 0f97abb..c560346 100644
--- a/common_audio/audio_converter.cc
+++ b/common_audio/audio_converter.cc
@@ -31,7 +31,7 @@
                 size_t dst_channels,
                 size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
-  ~CopyConverter() override{};
+  ~CopyConverter() override {}
 
   void Convert(const float* const* src,
                size_t src_size,
@@ -52,7 +52,7 @@
                  size_t dst_channels,
                  size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
-  ~UpmixConverter() override{};
+  ~UpmixConverter() override {}
 
   void Convert(const float* const* src,
                size_t src_size,
@@ -74,7 +74,7 @@
                    size_t dst_channels,
                    size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
-  ~DownmixConverter() override{};
+  ~DownmixConverter() override {}
 
   void Convert(const float* const* src,
                size_t src_size,
@@ -103,7 +103,7 @@
       resamplers_.push_back(std::unique_ptr<PushSincResampler>(
           new PushSincResampler(src_frames, dst_frames)));
   }
-  ~ResampleConverter() override{};
+  ~ResampleConverter() override {}
 
   void Convert(const float* const* src,
                size_t src_size,
@@ -132,7 +132,7 @@
           std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
               (*it)->dst_frames(), (*it)->dst_channels())));
   }
-  ~CompositionConverter() override{};
+  ~CompositionConverter() override {}
 
   void Convert(const float* const* src,
                size_t src_size,
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index 4392911..dff7887 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -172,4 +172,4 @@
   return *video_engine_.get();
 }
 
-};  // namespace cricket
+}  // namespace cricket
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 6731d50..ff6ac01 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -29,7 +29,7 @@
 constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
 }  // namespace
 
-AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr){};
+AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr) {}
 
 AudioNetworkAdaptorImpl::Config::~Config() = default;
 
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index b970ae9..dc14620 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -40,9 +40,9 @@
     float OptimizePacketLossRate(float packet_loss_rate) const;
 
     // Getters for testing.
-    float min_packet_loss_rate() const { return min_packet_loss_rate_; };
-    float max_packet_loss_rate() const { return max_packet_loss_rate_; };
-    float slope() const { return slope_; };
+    float min_packet_loss_rate() const { return min_packet_loss_rate_; }
+    float max_packet_loss_rate() const { return max_packet_loss_rate_; }
+    float slope() const { return slope_; }
 
    private:
     const float min_packet_loss_rate_;
diff --git a/modules/audio_processing/aec3/aec_state.h b/modules/audio_processing/aec3/aec_state.h
index 0ba9e19..f511429 100644
--- a/modules/audio_processing/aec3/aec_state.h
+++ b/modules/audio_processing/aec3/aec_state.h
@@ -306,7 +306,7 @@
   class SaturationDetector {
    public:
     // Returns whether the echo is to be considered saturated.
-    bool SaturatedEcho() const { return saturated_echo_; };
+    bool SaturatedEcho() const { return saturated_echo_; }
 
     // Updates the detection decision based on new data.
     void Update(rtc::ArrayView<const float> x,
@@ -327,7 +327,7 @@
     explicit LegacySaturationDetector(const EchoCanceller3Config& config);
 
     // Returns whether the echo is to be considered saturated.
-    bool SaturatedEcho() const { return saturated_echo_; };
+    bool SaturatedEcho() const { return saturated_echo_; }
 
     // Resets the state of the detector.
     void Reset();
diff --git a/modules/audio_processing/agc2/agc2_common.cc b/modules/audio_processing/agc2/agc2_common.cc
index af943df..1107885 100644
--- a/modules/audio_processing/agc2/agc2_common.cc
+++ b/modules/audio_processing/agc2/agc2_common.cc
@@ -54,4 +54,4 @@
   constexpr float kDefaultExtraSaturationMarginDb = 2.f;
   return kDefaultExtraSaturationMarginDb;
 }
-};  // namespace webrtc
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/gain_applier.h b/modules/audio_processing/agc2/gain_applier.h
index 7f9f00e..d9aa19d 100644
--- a/modules/audio_processing/agc2/gain_applier.h
+++ b/modules/audio_processing/agc2/gain_applier.h
@@ -22,7 +22,7 @@
 
   void ApplyGain(AudioFrameView<float> signal);
   void SetGainFactor(float gain_factor);
-  float GetGainFactor() const { return current_gain_factor_; };
+  float GetGainFactor() const { return current_gain_factor_; }
 
  private:
   void Initialize(size_t samples_per_channel);
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index cc879c8..6d84750 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -54,7 +54,7 @@
 ApmDataDumper::ApmDataDumper(int instance_index)
     : instance_index_(instance_index) {}
 #else
-ApmDataDumper::ApmDataDumper(int instance_index){};
+ApmDataDumper::ApmDataDumper(int instance_index) {}
 #endif
 
 ApmDataDumper::~ApmDataDumper() = default;
diff --git a/modules/desktop_capture/desktop_geometry.h b/modules/desktop_capture/desktop_geometry.h
index 78a9230..91608f0 100644
--- a/modules/desktop_capture/desktop_geometry.h
+++ b/modules/desktop_capture/desktop_geometry.h
@@ -136,7 +136,7 @@
 
   // Adds (dx, dy) to the position of the rectangle.
   void Translate(int32_t dx, int32_t dy);
-  void Translate(DesktopVector d) { Translate(d.x(), d.y()); };
+  void Translate(DesktopVector d) { Translate(d.x(), d.y()); }
 
   // Enlarges current DesktopRect by subtracting |left_offset| and |top_offset|
   // from |left_| and |top_|, and adding |right_offset| and |bottom_offset| to
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 90af86d..8d2a8f5 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -109,7 +109,7 @@
   virtual bool GetApplyRotation() = 0;
 
  protected:
-  ~VideoCaptureModule() override{};
+  ~VideoCaptureModule() override {}
 };
 
 }  // namespace webrtc
diff --git a/modules/video_capture/video_capture_defines.h b/modules/video_capture/video_capture_defines.h
index 3987c03..f69d03e 100644
--- a/modules/video_capture/video_capture_defines.h
+++ b/modules/video_capture/video_capture_defines.h
@@ -35,7 +35,7 @@
     maxFPS = 0;
     videoType = VideoType::kUnknown;
     interlaced = false;
-  };
+  }
   bool operator!=(const VideoCaptureCapability& other) const {
     if (width != other.width)
       return true;
diff --git a/modules/video_coding/codecs/vp8/libvpx_interface.cc b/modules/video_coding/codecs/vp8/libvpx_interface.cc
index 8f6a3a6..6b39053 100644
--- a/modules/video_coding/codecs/vp8/libvpx_interface.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_interface.cc
@@ -141,7 +141,7 @@
         RTC_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
     }
     return VPX_CODEC_ERROR;
-  };
+  }
 
   vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
                                 vp8e_enc_control_id ctrl_id,
@@ -153,7 +153,7 @@
         RTC_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
     }
     return VPX_CODEC_ERROR;
-  };
+  }
 
   vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
                                 vp8e_enc_control_id ctrl_id,
@@ -165,7 +165,7 @@
         RTC_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
     }
     return VPX_CODEC_ERROR;
-  };
+  }
 
   vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
                                 vp8e_enc_control_id ctrl_id,
@@ -177,7 +177,7 @@
         RTC_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
     }
     return VPX_CODEC_ERROR;
-  };
+  }
 
   vpx_codec_err_t codec_encode(vpx_codec_ctx_t* ctx,
                                const vpx_image_t* img,
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 5f07eb4..2358f1b 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -384,7 +384,7 @@
 
   int16_t network_cost() const { return network_cost_; }
 
-  void GetStunStats(absl::optional<StunStats>* stats) override{};
+  void GetStunStats(absl::optional<StunStats>* stats) override {}
 
  protected:
   enum { MSG_DESTROY_IF_DEAD = 0, MSG_FIRST_AVAILABLE };
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index fd74fde..2c04bd7 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -202,18 +202,18 @@
 
 BEGIN_SIGNALING_PROXY_MAP(RtpTransceiver)
 PROXY_SIGNALING_THREAD_DESTRUCTOR()
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type);
-PROXY_CONSTMETHOD0(absl::optional<std::string>, mid);
-PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender);
-PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver);
-PROXY_CONSTMETHOD0(bool, stopped);
-PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction);
-PROXY_METHOD1(void, SetDirection, RtpTransceiverDirection);
-PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction);
-PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction);
-PROXY_METHOD0(void, Stop);
-PROXY_METHOD1(void, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>);
-END_PROXY_MAP();
+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+PROXY_CONSTMETHOD0(absl::optional<std::string>, mid)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver)
+PROXY_CONSTMETHOD0(bool, stopped)
+PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction)
+PROXY_METHOD1(void, SetDirection, RtpTransceiverDirection)
+PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction)
+PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction)
+PROXY_METHOD0(void, Stop)
+PROXY_METHOD1(void, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>)
+END_PROXY_MAP()
 
 }  // namespace webrtc
 
diff --git a/rtc_base/async_packet_socket.cc b/rtc_base/async_packet_socket.cc
index 12afbeb..a708fae 100644
--- a/rtc_base/async_packet_socket.cc
+++ b/rtc_base/async_packet_socket.cc
@@ -42,4 +42,4 @@
   }
 }
 
-};  // namespace rtc
+}  // namespace rtc
diff --git a/rtc_base/async_resolver_interface.cc b/rtc_base/async_resolver_interface.cc
index c8d6ab8..ff8c87e 100644
--- a/rtc_base/async_resolver_interface.cc
+++ b/rtc_base/async_resolver_interface.cc
@@ -16,4 +16,4 @@
 
 AsyncResolverInterface::~AsyncResolverInterface() = default;
 
-};  // namespace rtc
+}  // namespace rtc
diff --git a/rtc_base/crypt_string.cc b/rtc_base/crypt_string.cc
index 3238255..ed0ff16 100644
--- a/rtc_base/crypt_string.cc
+++ b/rtc_base/crypt_string.cc
@@ -69,4 +69,4 @@
   memcpy(&dest->front(), password_.data(), password_.size());
 }
 
-};  // namespace rtc
+}  // namespace rtc
diff --git a/stats/rtc_stats.cc b/stats/rtc_stats.cc
index 4332fe6..f5139a7 100644
--- a/stats/rtc_stats.cc
+++ b/stats/rtc_stats.cc
@@ -154,79 +154,79 @@
                              false,
                              false,
                              rtc::ToString(value_),
-                             rtc::ToString(value_));
+                             rtc::ToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(int32_t,
                              kInt32,
                              false,
                              false,
                              rtc::ToString(value_),
-                             rtc::ToString(value_));
+                             rtc::ToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(uint32_t,
                              kUint32,
                              false,
                              false,
                              rtc::ToString(value_),
-                             rtc::ToString(value_));
+                             rtc::ToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(int64_t,
                              kInt64,
                              false,
                              false,
                              rtc::ToString(value_),
-                             ToStringAsDouble(value_));
+                             ToStringAsDouble(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(uint64_t,
                              kUint64,
                              false,
                              false,
                              rtc::ToString(value_),
-                             ToStringAsDouble(value_));
+                             ToStringAsDouble(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(double,
                              kDouble,
                              false,
                              false,
                              rtc::ToString(value_),
-                             ToStringAsDouble(value_));
-WEBRTC_DEFINE_RTCSTATSMEMBER(std::string, kString, false, true, value_, value_);
+                             ToStringAsDouble(value_))
+WEBRTC_DEFINE_RTCSTATSMEMBER(std::string, kString, false, true, value_, value_)
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<bool>,
                              kSequenceBool,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToString(value_));
+                             VectorToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<int32_t>,
                              kSequenceInt32,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToString(value_));
+                             VectorToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<uint32_t>,
                              kSequenceUint32,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToString(value_));
+                             VectorToString(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<int64_t>,
                              kSequenceInt64,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToStringAsDouble(value_));
+                             VectorToStringAsDouble(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<uint64_t>,
                              kSequenceUint64,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToStringAsDouble(value_));
+                             VectorToStringAsDouble(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<double>,
                              kSequenceDouble,
                              true,
                              false,
                              VectorToString(value_),
-                             VectorToStringAsDouble(value_));
+                             VectorToStringAsDouble(value_))
 WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<std::string>,
                              kSequenceString,
                              true,
                              false,
                              VectorOfStringsToString(value_),
-                             VectorOfStringsToString(value_));
+                             VectorOfStringsToString(value_))
 
 }  // namespace webrtc
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index 473a3f3..14764d7 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -56,7 +56,7 @@
     &fingerprint,
     &fingerprint_algorithm,
     &base64_certificate,
-    &issuer_certificate_id);
+    &issuer_certificate_id)
 // clang-format on
 
 RTCCertificateStats::RTCCertificateStats(const std::string& id,
@@ -86,7 +86,7 @@
     &clock_rate,
     &channels,
     &sdp_fmtp_line,
-    &implementation);
+    &implementation)
 // clang-format on
 
 RTCCodecStats::RTCCodecStats(const std::string& id, int64_t timestamp_us)
@@ -121,7 +121,7 @@
     &messages_sent,
     &bytes_sent,
     &messages_received,
-    &bytes_received);
+    &bytes_received)
 // clang-format on
 
 RTCDataChannelStats::RTCDataChannelStats(const std::string& id,
@@ -177,7 +177,7 @@
     &consent_requests_received,
     &consent_requests_sent,
     &consent_responses_received,
-    &consent_responses_sent);
+    &consent_responses_sent)
 // clang-format on
 
 RTCIceCandidatePairStats::RTCIceCandidatePairStats(const std::string& id,
@@ -254,7 +254,7 @@
     &candidate_type,
     &priority,
     &url,
-    &deleted);
+    &deleted)
 // clang-format on
 
 RTCIceCandidateStats::RTCIceCandidateStats(const std::string& id,
@@ -333,7 +333,7 @@
 // clang-format off
 WEBRTC_RTCSTATS_IMPL(RTCMediaStreamStats, RTCStats, "stream",
     &stream_identifier,
-    &track_ids);
+    &track_ids)
 // clang-format on
 
 RTCMediaStreamStats::RTCMediaStreamStats(const std::string& id,
@@ -387,7 +387,7 @@
                      &total_freezes_duration,
                      &total_pauses_duration,
                      &total_frames_duration,
-                     &sum_squared_frame_durations);
+                     &sum_squared_frame_durations)
 // clang-format on
 
 RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(const std::string& id,
@@ -480,7 +480,7 @@
 // clang-format off
 WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
     &data_channels_opened,
-    &data_channels_closed);
+    &data_channels_closed)
 // clang-format on
 
 RTCPeerConnectionStats::RTCPeerConnectionStats(const std::string& id,
@@ -515,7 +515,7 @@
     &pli_count,
     &nack_count,
     &sli_count,
-    &qp_sum);
+    &qp_sum)
 // clang-format on
 
 RTCRTPStreamStats::RTCRTPStreamStats(const std::string& id,
@@ -575,7 +575,7 @@
     &burst_discard_rate,
     &gap_loss_rate,
     &gap_discard_rate,
-    &frames_decoded);
+    &frames_decoded)
 // clang-format on
 
 RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(const std::string& id,
@@ -632,7 +632,7 @@
     &packets_sent,
     &bytes_sent,
     &target_bitrate,
-    &frames_encoded);
+    &frames_encoded)
 // clang-format on
 
 RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(const std::string& id,
@@ -665,7 +665,7 @@
     &dtls_state,
     &selected_candidate_pair_id,
     &local_certificate_id,
-    &remote_certificate_id);
+    &remote_certificate_id)
 // clang-format on
 
 RTCTransportStats::RTCTransportStats(const std::string& id,
diff --git a/stats/test/rtc_test_stats.cc b/stats/test/rtc_test_stats.cc
index d5576e6..d8bcbb1 100644
--- a/stats/test/rtc_test_stats.cc
+++ b/stats/test/rtc_test_stats.cc
@@ -30,7 +30,7 @@
                      &m_sequence_int64,
                      &m_sequence_uint64,
                      &m_sequence_double,
-                     &m_sequence_string);
+                     &m_sequence_string)
 
 RTCTestStats::RTCTestStats(const std::string& id, int64_t timestamp_us)
     : RTCStats(id, timestamp_us),
diff --git a/system_wrappers/include/clock.h b/system_wrappers/include/clock.h
index 3d292c8..b2c6c5f 100644
--- a/system_wrappers/include/clock.h
+++ b/system_wrappers/include/clock.h
@@ -83,6 +83,6 @@
   std::unique_ptr<RWLockWrapper> lock_;
 };
 
-};  // namespace webrtc
+}  // namespace webrtc
 
 #endif  // SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
diff --git a/system_wrappers/source/clock.cc b/system_wrappers/source/clock.cc
index a0acf40..f69d13c 100644
--- a/system_wrappers/source/clock.cc
+++ b/system_wrappers/source/clock.cc
@@ -271,4 +271,4 @@
   time_us_ += microseconds;
 }
 
-};  // namespace webrtc
+}  // namespace webrtc
diff --git a/test/fake_encoder.cc b/test/fake_encoder.cc
index 85e6151..f4645ad 100644
--- a/test/fake_encoder.cc
+++ b/test/fake_encoder.cc
@@ -45,7 +45,7 @@
   payload[3] = (counter & 0xFF000000) >> 24;
 }
 
-};  // namespace
+}  // namespace
 
 FakeEncoder::FakeEncoder(Clock* clock)
     : clock_(clock),