admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Comment unused variables in implemented functions 12\n

Browse Source 
Bug: webrtc:370878648
Change-Id: Ia9b1db4f6c393a016c3769cd57c540704e9ca4f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366526
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43329}
This commit is contained in:
Dor Hen 2024-10-29 14:48:24 +02:00 committed by WebRTC LUCI CQ
parent a154b73097
commit c118881416
29 changed files with 85 additions and 78 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
api/peer_connection_interface.cc
View File

@ -115,7 +115,7 @@ RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities(
}
RtpCapabilities PeerConnectionFactoryInterface::GetRtpReceiverCapabilities(
cricket::MediaType kind) const {
cricket::MediaType /* kind */) const {
return {};
}

21
media/base/fake_media_engine.cc
View File

@ -38,12 +38,12 @@ FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
}
}
void FakeVoiceMediaReceiveChannel::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 FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}
@ -141,12 +141,13 @@ std::optional<int> FakeVoiceMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
}
return std::nullopt;
}
bool FakeVoiceMediaReceiveChannel::GetStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) {
bool FakeVoiceMediaReceiveChannel::GetStats(
VoiceMediaReceiveInfo* /* info */,
bool /* get_and_clear_legacy_stats */) {
return false;
}
void FakeVoiceMediaReceiveChannel::SetRawAudioSink(
uint32_t ssrc,
uint32_t /* ssrc */,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(sink);
}
@ -155,7 +156,7 @@ void FakeVoiceMediaReceiveChannel::SetDefaultRawAudioSink(
sink_ = std::move(sink);
}
std::vector<webrtc::RtpSource> FakeVoiceMediaReceiveChannel::GetSources(
uint32_t ssrc) const {
uint32_t /* ssrc */) const {
return std::vector<webrtc::RtpSource>();
}
bool FakeVoiceMediaReceiveChannel::SetRecvCodecs(

20
media/base/fake_media_engine.h
View File

@ -581,7 +581,7 @@ class FakeVoiceMediaSendChannel
void SetReceiveNonSenderRttEnabled(bool /* enabled */) {}
bool SendCodecHasNack() const override { return false; }
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
absl::AnyInvocable<void()> /* callback */) override {}
std::optional<Codec> GetSendCodec() const override;
bool GetStats(VoiceMediaSendInfo* stats) override;
@ -658,7 +658,7 @@ class FakeVideoMediaReceiveChannel
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool HasSink(uint32_t ssrc) const;
void SetReceive(bool receive) override {}
void SetReceive(bool /* receive */) override {}
bool HasSource(uint32_t ssrc) const;
bool AddRecvStream(const StreamParams& sp) override;
@ -675,13 +675,14 @@ class FakeVideoMediaReceiveChannel
override;
void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
void RequestRecvKeyFrame(uint32_t ssrc) override;
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
std::optional<int> rtx_time) override {}
void SetReceiverFeedbackParameters(
bool /* lntf_enabled */,
bool /* nack_enabled */,
webrtc::RtcpMode /* rtcp_mode */,
std::optional<int> /* rtx_time */) override {}
bool GetStats(VideoMediaReceiveInfo* info) override;
bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
bool AddDefaultRecvStreamForTesting(const StreamParams& /* sp */) override {
RTC_CHECK_NOTREACHED();
return false;
}
@ -742,9 +743,10 @@ class FakeVideoMediaSendChannel
return webrtc::RtcpMode::kCompound;
}
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
absl::AnyInvocable<void()> /* callback */) override {}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {}
absl::AnyInvocable<void(const std::set<uint32_t>&)> /* callback */)
override {}
bool SendCodecHasLntf() const override { return false; }
bool SendCodecHasNack() const override { return false; }

8
media/engine/fake_webrtc_call.cc
View File

@ -137,7 +137,7 @@ void FakeAudioReceiveStream::SetFrameDecryptor(
}
webrtc::AudioReceiveStreamInterface::Stats FakeAudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
bool /* get_and_clear_legacy_stats */) const {
return stats_;
}
@ -349,7 +349,7 @@ void FakeVideoSendStream::Stop() {
}
void FakeVideoSendStream::AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) {}
rtc::scoped_refptr<webrtc::Resource> /* resource */) {}
std::vector<rtc::scoped_refptr<webrtc::Resource>>
FakeVideoSendStream::GetAdaptationResources() {
@ -637,7 +637,7 @@ void FakeCall::DestroyFlexfecReceiveStream(
}
void FakeCall::AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) {}
rtc::scoped_refptr<webrtc::Resource> /* resource */) {}
webrtc::PacketReceiver* FakeCall::Receiver() {
return this;
@ -728,7 +728,7 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
}
void FakeCall::OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) {}
int /* transport_overhead_per_packet */) {}
void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
uint32_t local_ssrc) {

4
media/engine/fake_webrtc_call.h
View File

@ -465,7 +465,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
void SetStats(const webrtc::Call::Stats& stats);
void SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) override {}
const webrtc::BitrateSettings& /* preferences */) override {}
const webrtc::FieldTrialsView& trials() const override {
return env_.field_trials();
}
@ -500,7 +500,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
webrtc::PacketReceiver* Receiver() override;
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override {}
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer /* packet */) override {}
void DeliverRtpPacket(
webrtc::MediaType media_type,

2
modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
View File

@ -50,7 +50,7 @@ class PowerRatioEstimator : public LappedTransform::Callback {
size_t num_input_channels,
size_t /* num_freq_bins */,
size_t /* num_output_channels */,
std::complex<float>* const* output) override {
std::complex<float>* const* /* output */) override {
float low_pow = 0.f;
float high_pow = 0.f;
for (size_t i = 0u; i < num_input_channels; ++i) {

2
modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -187,7 +187,7 @@ void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
block_length_ms * sample_rate_khz * channels_));
}
void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* /* encoder */,
opus_int32 expect,
int32_t set) {
opus_int32 bandwidth;

4
modules/audio_coding/codecs/opus/test/blocker_unittest.cc
View File

@ -22,7 +22,7 @@ class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
size_t num_input_channels,
size_t /* num_input_channels */,
size_t num_output_channels,
float* const* output) override {
for (size_t i = 0; i < num_output_channels; ++i) {
@ -38,7 +38,7 @@ class CopyBlockerCallback : public webrtc::BlockerCallback {
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
size_t num_input_channels,
size_t /* num_input_channels */,
size_t num_output_channels,
float* const* output) override {
for (size_t i = 0; i < num_output_channels; ++i) {

2
modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
View File

@ -50,7 +50,7 @@ class FftCheckerCallback : public webrtc::LappedTransform::Callback {
size_t in_channels,
size_t frames,
size_t out_channels,
complex<float>* const* out_block) override {
complex<float>* const* /* out_block */) override {
RTC_CHECK_EQ(in_channels, out_channels);
size_t full_length = (frames - 1) * 2;

2
modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
View File

@ -61,7 +61,7 @@ std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
samples_per_ms);
}
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
int AudioDecoderPcm16B::PacketDuration(const uint8_t* /* encoded */,
size_t encoded_len) const {
// Two encoded byte per sample per channel.
return static_cast<int>(encoded_len / (2 * Channels()));

12
modules/audio_coding/include/audio_coding_module.h
View File

@ -39,17 +39,17 @@ class AudioPacketizationCallback {
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) {
int64_t /* absolute_capture_timestamp_ms */) {
// TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
// pure virtual.
return SendData(frame_type, payload_type, timestamp, payload_data,
payload_len_bytes);
}
virtual int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) {
virtual int32_t SendData(AudioFrameType /* frame_type */,
uint8_t /* payload_type */,
uint32_t /* timestamp */,
const uint8_t* /* payload_data */,
size_t /* payload_len_bytes */) {
RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
return -1;
}

4
modules/audio_coding/neteq/background_noise.cc
View File

@ -120,8 +120,8 @@ bool BackgroundNoise::Update(const AudioMultiVector& sync_buffer) {
void BackgroundNoise::GenerateBackgroundNoise(
rtc::ArrayView<const int16_t> random_vector,
size_t channel,
int mute_slope,
bool too_many_expands,
int /* mute_slope */,
bool /* too_many_expands */,
size_t num_noise_samples,
int16_t* buffer) {
constexpr size_t kNoiseLpcOrder = kMaxLpcOrder;

2
modules/audio_coding/neteq/decision_logic.cc
View File

@ -114,7 +114,7 @@ void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
}
NetEq::Operation DecisionLogic::GetDecision(const NetEqStatus& status,
bool* reset_decoder) {
bool* /* reset_decoder */) {
prev_time_scale_ = prev_time_scale_ && IsTimestretch(status.last_mode);
if (prev_time_scale_) {
timescale_countdown_ = tick_timer_->GetNewCountdown(kMinTimescaleInterval);

2
modules/audio_coding/neteq/decision_logic.h
View File

@ -64,7 +64,7 @@ class DecisionLogic : public NetEqController {
NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status,
bool* reset_decoder) override;
void ExpandDecision(NetEq::Operation operation) override {}
void ExpandDecision(NetEq::Operation /* operation */) override {}
// Adds `value` to `sample_memory_`.
void AddSampleMemory(int32_t value) override { sample_memory_ += value; }

2
modules/audio_coding/neteq/expand_unittest.cc
View File

@ -56,7 +56,7 @@ class FakeStatisticsCalculator : public StatisticsCalculator {
FakeStatisticsCalculator(TickTimer* tick_timer)
: StatisticsCalculator(tick_timer) {}
void LogDelayedPacketOutageEvent(int num_samples, int fs_hz) override {
void LogDelayedPacketOutageEvent(int num_samples, int /* fs_hz */) override {
last_outage_duration_samples_ = num_samples;
}

8
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -494,7 +494,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
CountingSamplesDecoder() : next_value_(1) {}
// Produce as many samples as input bytes (`encoded_len`).
int DecodeInternal(const uint8_t* encoded,
int DecodeInternal(const uint8_t* /* encoded */,
size_t encoded_len,
int /* sample_rate_hz */,
int16_t* decoded,
@ -1690,7 +1690,7 @@ TEST_F(NetEqImplTest, NoCrashWith1000Channels) {
EXPECT_CALL(*mock_decoder_database_, GetActiveDecoder())
.WillRepeatedly(Return(decoder));
EXPECT_CALL(*mock_decoder_database_, SetActiveDecoder(_, _))
.WillOnce(Invoke([](uint8_t rtp_payload_type, bool* new_decoder) {
.WillOnce(Invoke([](uint8_t /* rtp_payload_type */, bool* new_decoder) {
*new_decoder = true;
return 0;
}));
@ -1803,8 +1803,8 @@ class Decoder120ms : public AudioDecoder {
next_value_(1),
speech_type_(speech_type) {}
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int DecodeInternal(const uint8_t* /* encoded */,
size_t /* encoded_len */,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override {

19
modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -77,7 +77,7 @@ class MockAudioDecoder final : public AudioDecoder {
const size_t num_channels_;
};
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
std::vector<ParseResult> ParsePayload(rtc::Buffer&& /* payload */,
uint32_t timestamp) override {
std::vector<ParseResult> results;
if (fec_enabled_) {
@ -91,14 +91,15 @@ class MockAudioDecoder final : public AudioDecoder {
return results;
}
int PacketDuration(const uint8_t* encoded,
size_t encoded_len) const override {
int PacketDuration(const uint8_t* /* encoded */,
size_t /* encoded_len */) const override {
ADD_FAILURE() << "Since going through ParsePayload, PacketDuration should "
"never get called.";
return kPacketDuration;
}
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override {
bool PacketHasFec(const uint8_t* /* encoded */,
size_t /* encoded_len */) const override {
ADD_FAILURE() << "Since going through ParsePayload, PacketHasFec should "
"never get called.";
return fec_enabled_;
@ -113,11 +114,11 @@ class MockAudioDecoder final : public AudioDecoder {
bool fec_enabled() const { return fec_enabled_; }
protected:
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override {
int DecodeInternal(const uint8_t* /* encoded */,
size_t /* encoded_len */,
int /* sample_rate_hz */,
int16_t* /* decoded */,
SpeechType* /* speech_type */) override {
ADD_FAILURE() << "Since going through ParsePayload, DecodeInternal should "
"never get called.";
return -1;

4
modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
View File

@ -52,9 +52,9 @@ class NetEqPcm16bQualityTest : public NetEqQualityTest {
}
int EncodeBlock(int16_t* in_data,
size_t block_size_samples,
size_t /* block_size_samples */,
rtc::Buffer* payload,
size_t max_bytes) override {
size_t /* max_bytes */) override {
const size_t kFrameSizeSamples = 480; // Samples per 10 ms.
size_t encoded_samples = 0;
uint32_t dummy_timestamp = 0;

4
modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
View File

@ -51,9 +51,9 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
}
int EncodeBlock(int16_t* in_data,
size_t block_size_samples,
size_t /* block_size_samples */,
rtc::Buffer* payload,
size_t max_bytes) override {
size_t /* max_bytes */) override {
const size_t kFrameSizeSamples = 80; // Samples per 10 ms.
size_t encoded_samples = 0;
uint32_t dummy_timestamp = 0;

3
modules/audio_coding/neteq/tools/audio_sink.cc
View File

@ -18,7 +18,8 @@ bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) {
right_sink_->WriteArray(audio, num_samples);
}
bool VoidAudioSink::WriteArray(const int16_t* audio, size_t num_samples) {
bool VoidAudioSink::WriteArray(const int16_t* /* audio */,
size_t /* num_samples */) {
return true;
}

2
modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
View File

@ -86,7 +86,7 @@ void PrintDelays(const NetEqDelayAnalyzer::Delays& delays,
void NetEqDelayAnalyzer::AfterInsertPacket(
const test::NetEqInput::PacketData& packet,
NetEq* neteq) {
NetEq* /* neteq */) {
data_.insert(
std::make_pair(packet.header.timestamp, TimingData(packet.time_ms)));
ssrcs_.insert(packet.header.ssrc);

4
modules/audio_coding/neteq/tools/neteq_quality_test.cc
View File

@ -259,13 +259,13 @@ NetEqQualityTest::~NetEqQualityTest() {
log_file_.close();
}
bool NoLoss::Lost(int now_ms) {
bool NoLoss::Lost(int /* now_ms */) {
return false;
}
UniformLoss::UniformLoss(double loss_rate) : loss_rate_(loss_rate) {}
bool UniformLoss::Lost(int now_ms) {
bool UniformLoss::Lost(int /* now_ms */) {
int drop_this = rand();
return (drop_this < loss_rate_ * RAND_MAX);
}

3
modules/audio_coding/neteq/tools/neteq_test.h
View File

@ -33,7 +33,8 @@ namespace test {
class NetEqTestErrorCallback {
public:
virtual ~NetEqTestErrorCallback() = default;
virtual void OnInsertPacketError(const NetEqInput::PacketData& packet) {}
virtual void OnInsertPacketError(const NetEqInput::PacketData& /* packet */) {
}
virtual void OnGetAudioError() {}
};

4
modules/audio_coding/neteq/tools/rtp_encode.cc
View File

@ -107,12 +107,12 @@ class Packetizer : public AudioPacketizationCallback {
ssrc_(ssrc),
timestamp_rate_hz_(timestamp_rate_hz) {}
int32_t SendData(AudioFrameType frame_type,
int32_t SendData(AudioFrameType /* frame_type */,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
int64_t /* absolute_capture_timestamp_ms */) override {
if (payload_len_bytes == 0) {
return 0;
}

2
modules/audio_coding/test/Channel.cc
View File

@ -22,7 +22,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
int64_t /* absolute_capture_timestamp_ms */) {
RTPHeader rtp_header;
int32_t status;
size_t payloadDataSize = payloadSize;

13
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -39,12 +39,13 @@ TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
TestPacketization::~TestPacketization() {}
int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
int32_t TestPacketization::SendData(
const AudioFrameType /* frameType */,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
int64_t /* absolute_capture_timestamp_ms */) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
return 1;

2
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -68,7 +68,7 @@ int32_t TestPack::SendData(AudioFrameType frame_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms) {
int64_t /* absolute_capture_timestamp_ms */) {
RTPHeader rtp_header;
int32_t status;

2
modules/audio_coding/test/TestStereo.cc
View File

@ -47,7 +47,7 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
const uint32_t timestamp,
const uint8_t* payload_data,
const size_t payload_size,
int64_t absolute_capture_timestamp_ms) {
int64_t /* absolute_capture_timestamp_ms */) {
RTPHeader rtp_header;
int32_t status = 0;

4
modules/audio_device/audio_device_data_observer.cc
View File

@ -122,8 +122,8 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
return res;
}
void PullRenderData(int bits_per_sample,
int sample_rate,
void PullRenderData(int /* bits_per_sample */,
int /* sample_rate */,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,