From c11888141631bb7744ceff9814b7415f551418ce Mon Sep 17 00:00:00 2001 From: Dor Hen Date: Tue, 29 Oct 2024 14:48:24 +0200 Subject: [PATCH] Comment unused variables in implemented functions 12\n Bug: webrtc:370878648 Change-Id: Ia9b1db4f6c393a016c3769cd57c540704e9ca4f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366526 Reviewed-by: Danil Chapovalov Commit-Queue: Dor Hen Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#43329} --- api/peer_connection_interface.cc | 2 +- media/base/fake_media_engine.cc | 21 ++++++++++--------- media/base/fake_media_engine.h | 20 ++++++++++-------- media/engine/fake_webrtc_call.cc | 8 +++---- media/engine/fake_webrtc_call.h | 4 ++-- .../codecs/opus/opus_bandwidth_unittest.cc | 2 +- .../audio_coding/codecs/opus/opus_unittest.cc | 2 +- .../codecs/opus/test/blocker_unittest.cc | 4 ++-- .../opus/test/lapped_transform_unittest.cc | 2 +- .../codecs/pcm16b/audio_decoder_pcm16b.cc | 2 +- .../include/audio_coding_module.h | 12 +++++------ .../audio_coding/neteq/background_noise.cc | 4 ++-- modules/audio_coding/neteq/decision_logic.cc | 2 +- modules/audio_coding/neteq/decision_logic.h | 2 +- modules/audio_coding/neteq/expand_unittest.cc | 2 +- .../audio_coding/neteq/neteq_impl_unittest.cc | 8 +++---- .../neteq/neteq_network_stats_unittest.cc | 19 +++++++++-------- .../neteq/test/neteq_pcm16b_quality_test.cc | 4 ++-- .../neteq/test/neteq_pcmu_quality_test.cc | 4 ++-- .../audio_coding/neteq/tools/audio_sink.cc | 3 ++- .../neteq/tools/neteq_delay_analyzer.cc | 2 +- .../neteq/tools/neteq_quality_test.cc | 4 ++-- modules/audio_coding/neteq/tools/neteq_test.h | 3 ++- .../audio_coding/neteq/tools/rtp_encode.cc | 4 ++-- modules/audio_coding/test/Channel.cc | 2 +- modules/audio_coding/test/EncodeDecodeTest.cc | 13 ++++++------ modules/audio_coding/test/TestAllCodecs.cc | 2 +- modules/audio_coding/test/TestStereo.cc | 2 +- .../audio_device_data_observer.cc | 4 ++-- 29 files changed, 85 insertions(+), 78 deletions(-) diff --git a/api/peer_connection_interface.cc b/api/peer_connection_interface.cc index 99e75d9b6e..07c8660e39 100644 --- a/api/peer_connection_interface.cc +++ b/api/peer_connection_interface.cc @@ -115,7 +115,7 @@ RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities( } RtpCapabilities PeerConnectionFactoryInterface::GetRtpReceiverCapabilities( - cricket::MediaType kind) const { + cricket::MediaType /* kind */) const { return {}; } diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc index 6cdf64b3d5..957e44c2b0 100644 --- a/media/base/fake_media_engine.cc +++ b/media/base/fake_media_engine.cc @@ -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 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 /* absolute_capture_timestamp_ms */) {} void FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::OnClose() { source_ = nullptr; } @@ -141,12 +141,13 @@ std::optional 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 sink) { sink_ = std::move(sink); } @@ -155,7 +156,7 @@ void FakeVoiceMediaReceiveChannel::SetDefaultRawAudioSink( sink_ = std::move(sink); } std::vector FakeVoiceMediaReceiveChannel::GetSources( - uint32_t ssrc) const { + uint32_t /* ssrc */) const { return std::vector(); } bool FakeVoiceMediaReceiveChannel::SetRecvCodecs( diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h index 330a370f64..2c7cc8b853 100644 --- a/media/base/fake_media_engine.h +++ b/media/base/fake_media_engine.h @@ -581,7 +581,7 @@ class FakeVoiceMediaSendChannel void SetReceiveNonSenderRttEnabled(bool /* enabled */) {} bool SendCodecHasNack() const override { return false; } void SetSendCodecChangedCallback( - absl::AnyInvocable callback) override {} + absl::AnyInvocable /* callback */) override {} std::optional GetSendCodec() const override; bool GetStats(VoiceMediaSendInfo* stats) override; @@ -658,7 +658,7 @@ class FakeVideoMediaReceiveChannel rtc::VideoSinkInterface* 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 rtx_time) override {} + void SetReceiverFeedbackParameters( + bool /* lntf_enabled */, + bool /* nack_enabled */, + webrtc::RtcpMode /* rtcp_mode */, + std::optional /* 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 callback) override {} + absl::AnyInvocable /* callback */) override {} void SetSsrcListChangedCallback( - absl::AnyInvocable&)> callback) override {} + absl::AnyInvocable&)> /* callback */) + override {} bool SendCodecHasLntf() const override { return false; } bool SendCodecHasNack() const override { return false; } diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc index 894f69fedc..cb4bf58855 100644 --- a/media/engine/fake_webrtc_call.cc +++ b/media/engine/fake_webrtc_call.cc @@ -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 resource) {} + rtc::scoped_refptr /* resource */) {} std::vector> FakeVideoSendStream::GetAdaptationResources() { @@ -637,7 +637,7 @@ void FakeCall::DestroyFlexfecReceiveStream( } void FakeCall::AddAdaptationResource( - rtc::scoped_refptr resource) {} + rtc::scoped_refptr /* 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) { diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h index 38deaff1f6..0029b1d23c 100644 --- a/media/engine/fake_webrtc_call.h +++ b/media/engine/fake_webrtc_call.h @@ -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, diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc index b332326061..16332caac9 100644 --- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc @@ -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* const* output) override { + std::complex* const* /* output */) override { float low_pow = 0.f; float high_pow = 0.f; for (size_t i = 0u; i < num_input_channels; ++i) { diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index 4a9156ad58..b54412ac02 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -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; diff --git a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc index 9c8e789ba9..edca73980e 100644 --- a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc +++ b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc @@ -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) { diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc index 1003ed52e5..8273f8d7f8 100644 --- a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc +++ b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc @@ -50,7 +50,7 @@ class FftCheckerCallback : public webrtc::LappedTransform::Callback { size_t in_channels, size_t frames, size_t out_channels, - complex* const* out_block) override { + complex* const* /* out_block */) override { RTC_CHECK_EQ(in_channels, out_channels); size_t full_length = (frames - 1) * 2; diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc index 1e2b5db331..f87ad02c7f 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc +++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc @@ -61,7 +61,7 @@ std::vector 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(encoded_len / (2 * Channels())); diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h index 094ef0eb2b..0ecfc8a95a 100644 --- a/modules/audio_coding/include/audio_coding_module.h +++ b/modules/audio_coding/include/audio_coding_module.h @@ -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; } diff --git a/modules/audio_coding/neteq/background_noise.cc b/modules/audio_coding/neteq/background_noise.cc index 0c33dba47a..82fdc2a41d 100644 --- a/modules/audio_coding/neteq/background_noise.cc +++ b/modules/audio_coding/neteq/background_noise.cc @@ -120,8 +120,8 @@ bool BackgroundNoise::Update(const AudioMultiVector& sync_buffer) { void BackgroundNoise::GenerateBackgroundNoise( rtc::ArrayView 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; diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc index d79a4e1663..1244940e13 100644 --- a/modules/audio_coding/neteq/decision_logic.cc +++ b/modules/audio_coding/neteq/decision_logic.cc @@ -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); diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h index 4b974c55b7..26dd8a46ae 100644 --- a/modules/audio_coding/neteq/decision_logic.h +++ b/modules/audio_coding/neteq/decision_logic.h @@ -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; } diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc index 92bbf7d3d4..45ba3c1f61 100644 --- a/modules/audio_coding/neteq/expand_unittest.cc +++ b/modules/audio_coding/neteq/expand_unittest.cc @@ -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; } diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index 6d9d6feeb5..500e18501e 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -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 { diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc index b2e1338102..8a54170578 100644 --- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc +++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc @@ -77,7 +77,7 @@ class MockAudioDecoder final : public AudioDecoder { const size_t num_channels_; }; - std::vector ParsePayload(rtc::Buffer&& payload, + std::vector ParsePayload(rtc::Buffer&& /* payload */, uint32_t timestamp) override { std::vector 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; diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc index c3e160cb66..2cb7c89bdf 100644 --- a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc @@ -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; diff --git a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc index d22170c623..7b0b8f4260 100644 --- a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc @@ -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; diff --git a/modules/audio_coding/neteq/tools/audio_sink.cc b/modules/audio_coding/neteq/tools/audio_sink.cc index 7d7af7ef9f..656dda43a1 100644 --- a/modules/audio_coding/neteq/tools/audio_sink.cc +++ b/modules/audio_coding/neteq/tools/audio_sink.cc @@ -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; } diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc index 9e77457775..432b062bab 100644 --- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc +++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc @@ -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); diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc index 341d1f9e78..ac8bc56031 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -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); } diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h index 5f826e2fdc..874ece3f7e 100644 --- a/modules/audio_coding/neteq/tools/neteq_test.h +++ b/modules/audio_coding/neteq/tools/neteq_test.h @@ -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() {} }; diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc index c5707e155a..5e98036142 100644 --- a/modules/audio_coding/neteq/tools/rtp_encode.cc +++ b/modules/audio_coding/neteq/tools/rtp_encode.cc @@ -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; } diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc index 705f8a16a6..3879bde167 100644 --- a/modules/audio_coding/test/Channel.cc +++ b/modules/audio_coding/test/Channel.cc @@ -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; diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index 69838f968c..42ea935643 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -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; diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index b7beca4db4..0c56e6da7d 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -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; diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc index 3308cb0430..1b294764c0 100644 --- a/modules/audio_coding/test/TestStereo.cc +++ b/modules/audio_coding/test/TestStereo.cc @@ -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; diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc index 6f4a17793a..f0075651f1 100644 --- a/modules/audio_device/audio_device_data_observer.cc +++ b/modules/audio_device/audio_device_data_observer.cc @@ -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,