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Fix -Wextra-semi warnings.

Browse source 
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}
This commit is contained in:
Mirko Bonadei 2019-02-25 09:12:02 +01:00 committed by Commit Bot
parent 3812fa949a
commit c4dd730765
18 changed files with 74 additions and 74 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
api/test/fake_media_transport.h
View file

@ -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 {}

8
common_video/h264/sps_vui_rewriter_unittest.cc
View file

@ -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

6
examples/peerconnection/client/conductor.h
View file

@ -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 {}

2
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View file

@ -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));

32
modules/audio_coding/codecs/opus/opus_speed_test.cc
View file

@ -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[] = {

2
modules/audio_coding/neteq/tools/neteq_quality_test.h
View file

@ -34,7 +34,7 @@ enum LossModes {
class LossModel {
public:
virtual ~LossModel(){};
virtual ~LossModel() {}
virtual bool Lost(int now_ms) = 0;
};

2
modules/audio_coding/test/EncodeDecodeTest.h
View file

@ -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();

2
modules/audio_device/linux/audio_device_alsa_linux.cc
View file

@ -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;

4
modules/audio_device/linux/audio_device_alsa_linux.h
View file

@ -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;

4
modules/audio_processing/aec3/echo_canceller3_unittest.cc
View file

@ -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_;

2
modules/audio_processing/audio_processing_impl_unittest.cc
View file

@ -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

2
modules/video_coding/generic_encoder_unittest.cc
View file

@ -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_; }

6
pc/rtc_stats_collector_unittest.cc
View file

@ -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.

58
rtc_base/ssl_stream_adapter_unittest.cc
View file

@ -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) {

6
stats/rtc_stats_report_unittest.cc
View file

@ -26,7 +26,7 @@ class RTCTestStats1 : public RTCStats {
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 @@ class RTCTestStats2 : public RTCStats {
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 @@ class RTCTestStats3 : public RTCStats {
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);

4
stats/rtc_stats_unittest.cc
View file

@ -49,7 +49,7 @@ class RTCChildStats : public RTCStats {
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 @@ class RTCGrandChildStats : public RTCChildStats {
WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats,
RTCChildStats,
"grandchild-stats",
&grandchild_int);
&grandchild_int)
TEST(RTCStatsTest, RTCStatsAndMembers) {
RTCTestStats stats("testId", 42);

2
system_wrappers/source/rtp_to_ntp_estimator_unittest.cc
View file

@ -346,4 +346,4 @@ TEST(RtpToNtpTests, AveragesErrorOut) {
}
}
}; // namespace webrtc
} // namespace webrtc

4
test/scenario/network/traffic_route.cc
View file

@ -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.