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/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_; }