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 <[email protected]> Commit-Queue: Mirko Bonadei <[email protected]> Cr-Commit-Position: refs/heads/master@{#26831}
JumpingYang001 · Feb 25, 2019 · c4dd730 · c4dd730
1 parent 3812fa9
commit c4dd730
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Showing 18 changed files with 74 additions and 74 deletions.
diff --git 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
@@ -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
@@ -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<webrtc::RtpReceiverInterface> receiver,
       const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
@@ -74,9 +74,9 @@ class Conductor : public webrtc::PeerConnectionObserver,
       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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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<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
@@ -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
@@ -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
@@ -214,8 +214,8 @@ class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
   }
 
   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 @@ class RTCTestStats : public RTCStats {
   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.