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Comment unused variables in implemented functions 13\n
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Increased the number of errors the automation is fixing to 150 from
75 in this commit.

Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <[email protected]>
Reviewed-by: Danil Chapovalov <[email protected]>
Commit-Queue: Dor Hen <[email protected]>
Cr-Commit-Position: refs/heads/main@{#43337}
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Dor Hen authored and WebRTC LUCI CQ committed Oct 31, 2024
1 parent 297fe1a commit 3fa21c8
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Showing 53 changed files with 389 additions and 351 deletions.
4 changes: 2 additions & 2 deletions audio/voip/test/audio_egress_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -195,7 +195,7 @@ TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
constexpr int kExpected = 10;
rtc::Event event;
int rtp_count = 0;
auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
auto rtp_sent = [&](rtc::ArrayView<const uint8_t> /* packet */, Unused) {
if (++rtp_count == kExpected) {
event.Set();
}
Expand Down Expand Up @@ -296,7 +296,7 @@ TEST_F(AudioEgressTest, TestAudioInputLevelAndEnergyDuration) {
constexpr int kExpected = 6;
rtc::Event event;
int rtp_count = 0;
auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
auto rtp_sent = [&](rtc::ArrayView<const uint8_t> /* packet */, Unused) {
if (++rtp_count == kExpected) {
event.Set();
}
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73 changes: 38 additions & 35 deletions media/base/fake_media_engine.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -194,12 +194,12 @@ FakeVoiceMediaSendChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
}
}
void FakeVoiceMediaSendChannel::VoiceChannelAudioSink::OnData(
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
std::optional<int64_t> absolute_capture_timestamp_ms) {}
const void* /* audio_data */,
int /* bits_per_sample */,
int /* sample_rate */,
size_t /* number_of_channels */,
size_t /* number_of_frames */,
std::optional<int64_t> /* absolute_capture_timestamp_ms */) {}
void FakeVoiceMediaSendChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}
Expand Down Expand Up @@ -288,7 +288,7 @@ bool FakeVoiceMediaSendChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
*volume = output_scalings_[ssrc];
return true;
}
bool FakeVoiceMediaSendChannel::GetStats(VoiceMediaSendInfo* info) {
bool FakeVoiceMediaSendChannel::GetStats(VoiceMediaSendInfo* /* info */) {
return false;
}
bool FakeVoiceMediaSendChannel::SetSendCodecs(
Expand Down Expand Up @@ -388,8 +388,8 @@ bool FakeVideoMediaSendChannel::HasSource(uint32_t ssrc) const {
return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr;
}
void FakeVideoMediaSendChannel::FillBitrateInfo(
BandwidthEstimationInfo* bwe_info) {}
bool FakeVideoMediaSendChannel::GetStats(VideoMediaSendInfo* info) {
BandwidthEstimationInfo* /* bwe_info */) {}
bool FakeVideoMediaSendChannel::GetStats(VideoMediaSendInfo* /* info */) {
return false;
}
bool FakeVideoMediaSendChannel::SetSendCodecs(
Expand All @@ -412,8 +412,8 @@ bool FakeVideoMediaSendChannel::SetMaxSendBandwidth(int bps) {
return true;
}
void FakeVideoMediaSendChannel::GenerateSendKeyFrame(
uint32_t ssrc,
const std::vector<std::string>& rids) {}
uint32_t /* ssrc */,
const std::vector<std::string>& /* rids */) {}

FakeVideoMediaReceiveChannel::FakeVideoMediaReceiveChannel(
const VideoOptions& options,
Expand Down Expand Up @@ -457,7 +457,7 @@ bool FakeVideoMediaReceiveChannel::SetSink(
return true;
}
void FakeVideoMediaReceiveChannel::SetDefaultSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {}
rtc::VideoSinkInterface<webrtc::VideoFrame>* /* sink */) {}
bool FakeVideoMediaReceiveChannel::HasSink(uint32_t ssrc) const {
return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
}
Expand All @@ -481,7 +481,7 @@ bool FakeVideoMediaReceiveChannel::RemoveRecvStream(uint32_t ssrc) {
return true;
}
std::vector<webrtc::RtpSource> FakeVideoMediaReceiveChannel::GetSources(
uint32_t ssrc) const {
uint32_t /* ssrc */) const {
return {};
}
bool FakeVideoMediaReceiveChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
Expand Down Expand Up @@ -521,15 +521,16 @@ bool FakeVideoMediaReceiveChannel::SetMaxSendBandwidth(int bps) {
}

void FakeVideoMediaReceiveChannel::SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback) {}
uint32_t /* ssrc */,
std::function<void(const webrtc::RecordableEncodedFrame&)> /* callback */) {
}

void FakeVideoMediaReceiveChannel::ClearRecordableEncodedFrameCallback(
uint32_t ssrc) {}
uint32_t /* ssrc */) {}

void FakeVideoMediaReceiveChannel::RequestRecvKeyFrame(uint32_t ssrc) {}
void FakeVideoMediaReceiveChannel::RequestRecvKeyFrame(uint32_t /* ssrc */) {}

bool FakeVideoMediaReceiveChannel::GetStats(VideoMediaReceiveInfo* info) {
bool FakeVideoMediaReceiveChannel::GetStats(VideoMediaReceiveInfo* /* info */) {
return false;
}

Expand All @@ -543,11 +544,12 @@ rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const {
return rtc::scoped_refptr<webrtc::AudioState>();
}
std::unique_ptr<VoiceMediaSendChannelInterface>
FakeVoiceEngine::CreateSendChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
FakeVoiceEngine::CreateSendChannel(
webrtc::Call* call,
const MediaConfig& /* config */,
const AudioOptions& options,
const webrtc::CryptoOptions& /* crypto_options */,
webrtc::AudioCodecPairId /* codec_pair_id */) {
std::unique_ptr<FakeVoiceMediaSendChannel> ch =
std::make_unique<FakeVoiceMediaSendChannel>(options,
call->network_thread());
Expand All @@ -556,10 +558,10 @@ FakeVoiceEngine::CreateSendChannel(webrtc::Call* call,
std::unique_ptr<VoiceMediaReceiveChannelInterface>
FakeVoiceEngine::CreateReceiveChannel(
webrtc::Call* call,
const MediaConfig& config,
const MediaConfig& /* config */,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
const webrtc::CryptoOptions& /* crypto_options */,
webrtc::AudioCodecPairId /* codec_pair_id */) {
std::unique_ptr<FakeVoiceMediaReceiveChannel> ch =
std::make_unique<FakeVoiceMediaReceiveChannel>(options,
call->network_thread());
Expand All @@ -584,8 +586,8 @@ void FakeVoiceEngine::SetSendCodecs(const std::vector<Codec>& codecs) {
int FakeVoiceEngine::GetInputLevel() {
return 0;
}
bool FakeVoiceEngine::StartAecDump(webrtc::FileWrapper file,
int64_t max_size_bytes) {
bool FakeVoiceEngine::StartAecDump(webrtc::FileWrapper /* file */,
int64_t /* max_size_bytes */) {
return false;
}
std::optional<webrtc::AudioDeviceModule::Stats>
Expand Down Expand Up @@ -618,10 +620,11 @@ bool FakeVideoEngine::SetOptions(const VideoOptions& options) {
std::unique_ptr<VideoMediaSendChannelInterface>
FakeVideoEngine::CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const MediaConfig& /* config */,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
const webrtc::CryptoOptions& /* crypto_options */,
webrtc::
VideoBitrateAllocatorFactory* /* video_bitrate_allocator_factory */) {
if (fail_create_channel_) {
return nullptr;
}
Expand All @@ -634,9 +637,9 @@ FakeVideoEngine::CreateSendChannel(
std::unique_ptr<VideoMediaReceiveChannelInterface>
FakeVideoEngine::CreateReceiveChannel(
webrtc::Call* call,
const MediaConfig& config,
const MediaConfig& /* config */,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) {
const webrtc::CryptoOptions& /* crypto_options */) {
if (fail_create_channel_) {
return nullptr;
}
Expand All @@ -646,11 +649,11 @@ FakeVideoEngine::CreateReceiveChannel(
call->network_thread());
return ch;
}
std::vector<Codec> FakeVideoEngine::send_codecs(bool use_rtx) const {
std::vector<Codec> FakeVideoEngine::send_codecs(bool /* use_rtx */) const {
return send_codecs_;
}

std::vector<Codec> FakeVideoEngine::recv_codecs(bool use_rtx) const {
std::vector<Codec> FakeVideoEngine::recv_codecs(bool /* use_rtx */) const {
return recv_codecs_;
}

Expand Down
2 changes: 1 addition & 1 deletion media/engine/webrtc_video_engine_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -4128,7 +4128,7 @@ class Vp9SettingsTestWithFieldTrial

void VerifySettings(int num_spatial_layers,
int num_temporal_layers,
webrtc::InterLayerPredMode interLayerPred) {
webrtc::InterLayerPredMode /* interLayerPred */) {
cricket::VideoSenderParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(send_channel_->SetSenderParameters(parameters));
Expand Down
6 changes: 3 additions & 3 deletions media/engine/webrtc_voice_engine_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -239,11 +239,11 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {

class FakeAudioSink : public webrtc::AudioSinkInterface {
public:
void OnData(const Data& audio) override {}
void OnData(const Data& /* audio */) override {}
};

class FakeAudioSource : public cricket::AudioSource {
void SetSink(Sink* sink) override {}
void SetSink(Sink* /* sink */) override {}
};

class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
Expand Down Expand Up @@ -714,7 +714,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
}
}
void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info,
bool is_sending) {
bool /* is_sending */) {
const auto stats = GetAudioSendStreamStats();
EXPECT_EQ(info.ssrc(), stats.local_ssrc);
EXPECT_EQ(info.payload_bytes_sent, stats.payload_bytes_sent);
Expand Down
10 changes: 5 additions & 5 deletions modules/audio_device/audio_device_data_observer.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -124,11 +124,11 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {

void PullRenderData(int /* bits_per_sample */,
int /* sample_rate */,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {
size_t /* number_of_channels */,
size_t /* number_of_frames */,
void* /* audio_data */,
int64_t* /* elapsed_time_ms */,
int64_t* /* ntp_time_ms */) override {
RTC_DCHECK_NOTREACHED();
}

Expand Down
4 changes: 2 additions & 2 deletions modules/audio_device/audio_device_impl.h
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Original file line number Diff line number Diff line change
Expand Up @@ -159,8 +159,8 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {

int RestartPlayoutInternally() override { return -1; }
int RestartRecordingInternally() override { return -1; }
int SetPlayoutSampleRate(uint32_t sample_rate) override { return -1; }
int SetRecordingSampleRate(uint32_t sample_rate) override { return -1; }
int SetPlayoutSampleRate(uint32_t /* sample_rate */) override { return -1; }
int SetRecordingSampleRate(uint32_t /* sample_rate */) override { return -1; }

private:
PlatformType Platform() const;
Expand Down
14 changes: 7 additions & 7 deletions modules/audio_device/audio_device_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -369,11 +369,11 @@ class MockAudioTransport : public test::MockAudioTransport {
const size_t bytes_per_frame,
const size_t channels,
const uint32_t sample_rate,
const uint32_t total_delay_ms,
const int32_t clock_drift,
const uint32_t current_mic_level,
const bool typing_status,
uint32_t& new_mic_level) {
const uint32_t /* total_delay_ms */,
const int32_t /* clock_drift */,
const uint32_t /* current_mic_level */,
const bool /* typing_status */,
uint32_t& /* new_mic_level */) {
EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
// Store audio parameters once in the first callback. For all other
// callbacks, verify that the provided audio parameters are maintained and
Expand Down Expand Up @@ -412,8 +412,8 @@ class MockAudioTransport : public test::MockAudioTransport {
const uint32_t sample_rate,
void* audio_buffer,
size_t& samples_out,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
int64_t* /* elapsed_time_ms */,
int64_t* /* ntp_time_ms */) {
EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
// Store audio parameters once in the first callback. For all other
// callbacks, verify that the provided audio parameters are maintained and
Expand Down
37 changes: 19 additions & 18 deletions modules/audio_device/dummy/audio_device_dummy.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -157,72 +157,73 @@ int32_t AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& /* available */) {
return -1;
}

int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t volume) {
int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t /* volume */) {
return -1;
}

int32_t AudioDeviceDummy::MicrophoneVolume(uint32_t& volume) const {
int32_t AudioDeviceDummy::MicrophoneVolume(uint32_t& /* volume */) const {
return -1;
}

int32_t AudioDeviceDummy::MaxMicrophoneVolume(uint32_t& maxVolume) const {
int32_t AudioDeviceDummy::MaxMicrophoneVolume(uint32_t& /* maxVolume */) const {
return -1;
}

int32_t AudioDeviceDummy::MinMicrophoneVolume(uint32_t& minVolume) const {
int32_t AudioDeviceDummy::MinMicrophoneVolume(uint32_t& /* minVolume */) const {
return -1;
}

int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available) {
int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& /* available */) {
return -1;
}

int32_t AudioDeviceDummy::SetSpeakerMute(bool enable) {
int32_t AudioDeviceDummy::SetSpeakerMute(bool /* enable */) {
return -1;
}

int32_t AudioDeviceDummy::SpeakerMute(bool& enabled) const {
int32_t AudioDeviceDummy::SpeakerMute(bool& /* enabled */) const {
return -1;
}

int32_t AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available) {
int32_t AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& /* available */) {
return -1;
}

int32_t AudioDeviceDummy::SetMicrophoneMute(bool enable) {
int32_t AudioDeviceDummy::SetMicrophoneMute(bool /* enable */) {
return -1;
}

int32_t AudioDeviceDummy::MicrophoneMute(bool& enabled) const {
int32_t AudioDeviceDummy::MicrophoneMute(bool& /* enabled */) const {
return -1;
}

int32_t AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available) {
int32_t AudioDeviceDummy::StereoPlayoutIsAvailable(bool& /* available */) {
return -1;
}
int32_t AudioDeviceDummy::SetStereoPlayout(bool enable) {
int32_t AudioDeviceDummy::SetStereoPlayout(bool /* enable */) {
return -1;
}

int32_t AudioDeviceDummy::StereoPlayout(bool& enabled) const {
int32_t AudioDeviceDummy::StereoPlayout(bool& /* enabled */) const {
return -1;
}

int32_t AudioDeviceDummy::StereoRecordingIsAvailable(bool& available) {
int32_t AudioDeviceDummy::StereoRecordingIsAvailable(bool& /* available */) {
return -1;
}

int32_t AudioDeviceDummy::SetStereoRecording(bool enable) {
int32_t AudioDeviceDummy::SetStereoRecording(bool /* enable */) {
return -1;
}

int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const {
int32_t AudioDeviceDummy::StereoRecording(bool& /* enabled */) const {
return -1;
}

int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const {
int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& /* delayMS */) const {
return -1;
}

void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* /* audioBuffer */) {
}
} // namespace webrtc
6 changes: 3 additions & 3 deletions modules/audio_device/dummy/file_audio_device.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -400,7 +400,7 @@ int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
available = true;
return 0;
}
int32_t FileAudioDevice::SetStereoPlayout(bool enable) {
int32_t FileAudioDevice::SetStereoPlayout(bool /* enable */) {
return 0;
}

Expand All @@ -414,7 +414,7 @@ int32_t FileAudioDevice::StereoRecordingIsAvailable(bool& available) {
return 0;
}

int32_t FileAudioDevice::SetStereoRecording(bool enable) {
int32_t FileAudioDevice::SetStereoRecording(bool /* enable */) {
return 0;
}

Expand All @@ -423,7 +423,7 @@ int32_t FileAudioDevice::StereoRecording(bool& enabled) const {
return 0;
}

int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
int32_t FileAudioDevice::PlayoutDelay(uint16_t& /* delayMS */) const {
return 0;
}

Expand Down
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