diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index 46b351bf4d..de1984db6a 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -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();
     }
@@ -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();
     }
diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index 957e44c2b0..8c084815f6 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -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;
 }
@@ -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(
@@ -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(
@@ -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,
@@ -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;
 }
@@ -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,
@@ -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;
 }
 
@@ -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());
@@ -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());
@@ -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>
@@ -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;
   }
@@ -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;
   }
@@ -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_;
 }
 
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index e2153e8026..913560a492 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -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));
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index aa5a835a83..c3576f6b0b 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -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> {
@@ -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);
diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc
index f0075651f1..9f348d4656 100644
--- a/modules/audio_device/audio_device_data_observer.cc
+++ b/modules/audio_device/audio_device_data_observer.cc
@@ -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();
   }
 
diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h
index 1e84eda02b..41d68209e8 100644
--- a/modules/audio_device/audio_device_impl.h
+++ b/modules/audio_device/audio_device_impl.h
@@ -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;
diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc
index 8090cc5a37..1a43af3414 100644
--- a/modules/audio_device/audio_device_unittest.cc
+++ b/modules/audio_device/audio_device_unittest.cc
@@ -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
@@ -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
diff --git a/modules/audio_device/dummy/audio_device_dummy.cc b/modules/audio_device/dummy/audio_device_dummy.cc
index 551575cddb..bc7b95d881 100644
--- a/modules/audio_device/dummy/audio_device_dummy.cc
+++ b/modules/audio_device/dummy/audio_device_dummy.cc
@@ -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
diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc
index 6009138cc4..7fef65318d 100644
--- a/modules/audio_device/dummy/file_audio_device.cc
+++ b/modules/audio_device/dummy/file_audio_device.cc
@@ -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;
 }
 
@@ -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;
 }
 
@@ -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;
 }
 
diff --git a/modules/audio_device/include/audio_device_default.h b/modules/audio_device/include/audio_device_default.h
index fb146a9d89..29b4123cd0 100644
--- a/modules/audio_device/include/audio_device_default.h
+++ b/modules/audio_device/include/audio_device_default.h
@@ -25,50 +25,50 @@ class AudioDeviceModuleDefault : public T {
   AudioDeviceModuleDefault() {}
   virtual ~AudioDeviceModuleDefault() {}
 
-  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
+  int32_t RegisterAudioCallback(AudioTransport* /* audioCallback */) override {
     return 0;
   }
   int32_t Init() override { return 0; }
   int32_t InitSpeaker() override { return 0; }
-  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
+  int32_t SetPlayoutDevice(uint16_t /* index */) override { return 0; }
   int32_t SetPlayoutDevice(
-      AudioDeviceModule::WindowsDeviceType device) override {
+      AudioDeviceModule::WindowsDeviceType /* device */) override {
     return 0;
   }
-  int32_t SetStereoPlayout(bool enable) override { return 0; }
+  int32_t SetStereoPlayout(bool /* enable */) override { return 0; }
   int32_t StopPlayout() override { return 0; }
   int32_t InitMicrophone() override { return 0; }
-  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
+  int32_t SetRecordingDevice(uint16_t /* index */) override { return 0; }
   int32_t SetRecordingDevice(
-      AudioDeviceModule::WindowsDeviceType device) override {
+      AudioDeviceModule::WindowsDeviceType /* device */) override {
     return 0;
   }
-  int32_t SetStereoRecording(bool enable) override { return 0; }
+  int32_t SetStereoRecording(bool /* enable */) override { return 0; }
   int32_t StopRecording() override { return 0; }
 
   int32_t Terminate() override { return 0; }
 
   int32_t ActiveAudioLayer(
-      AudioDeviceModule::AudioLayer* audioLayer) const override {
+      AudioDeviceModule::AudioLayer* /* audioLayer */) const override {
     return 0;
   }
   bool Initialized() const override { return true; }
   int16_t PlayoutDevices() override { return 0; }
   int16_t RecordingDevices() override { return 0; }
-  int32_t PlayoutDeviceName(uint16_t index,
-                            char name[kAdmMaxDeviceNameSize],
-                            char guid[kAdmMaxGuidSize]) override {
+  int32_t PlayoutDeviceName(uint16_t /* index */,
+                            char /* name */[kAdmMaxDeviceNameSize],
+                            char /* guid */[kAdmMaxGuidSize]) override {
     return 0;
   }
-  int32_t RecordingDeviceName(uint16_t index,
-                              char name[kAdmMaxDeviceNameSize],
-                              char guid[kAdmMaxGuidSize]) override {
+  int32_t RecordingDeviceName(uint16_t /* index */,
+                              char /* name */[kAdmMaxDeviceNameSize],
+                              char /* guid */[kAdmMaxGuidSize]) override {
     return 0;
   }
-  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
+  int32_t PlayoutIsAvailable(bool* /* available */) override { return 0; }
   int32_t InitPlayout() override { return 0; }
   bool PlayoutIsInitialized() const override { return true; }
-  int32_t RecordingIsAvailable(bool* available) override { return 0; }
+  int32_t RecordingIsAvailable(bool* /* available */) override { return 0; }
   int32_t InitRecording() override { return 0; }
   bool RecordingIsInitialized() const override { return true; }
   int32_t StartPlayout() override { return 0; }
@@ -77,40 +77,52 @@ class AudioDeviceModuleDefault : public T {
   bool Recording() const override { return false; }
   bool SpeakerIsInitialized() const override { return true; }
   bool MicrophoneIsInitialized() const override { return true; }
-  int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
-  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
-  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
-  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
-  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
-  int32_t MicrophoneVolumeIsAvailable(bool* available) override { return 0; }
-  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
-  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
-  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
-  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
-  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
-  int32_t SetSpeakerMute(bool enable) override { return 0; }
-  int32_t SpeakerMute(bool* enabled) const override { return 0; }
-  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
-  int32_t SetMicrophoneMute(bool enable) override { return 0; }
-  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
+  int32_t SpeakerVolumeIsAvailable(bool* /* available */) override { return 0; }
+  int32_t SetSpeakerVolume(uint32_t /* volume */) override { return 0; }
+  int32_t SpeakerVolume(uint32_t* /* volume */) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t* /* maxVolume */) const override {
+    return 0;
+  }
+  int32_t MinSpeakerVolume(uint32_t* /* minVolume */) const override {
+    return 0;
+  }
+  int32_t MicrophoneVolumeIsAvailable(bool* /* available */) override {
+    return 0;
+  }
+  int32_t SetMicrophoneVolume(uint32_t /* volume */) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t* /* volume */) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t* /* maxVolume */) const override {
+    return 0;
+  }
+  int32_t MinMicrophoneVolume(uint32_t* /* minVolume */) const override {
+    return 0;
+  }
+  int32_t SpeakerMuteIsAvailable(bool* /* available */) override { return 0; }
+  int32_t SetSpeakerMute(bool /* enable */) override { return 0; }
+  int32_t SpeakerMute(bool* /* enabled */) const override { return 0; }
+  int32_t MicrophoneMuteIsAvailable(bool* /* available */) override {
+    return 0;
+  }
+  int32_t SetMicrophoneMute(bool /* enable */) override { return 0; }
+  int32_t MicrophoneMute(bool* /* enabled */) const override { return 0; }
   int32_t StereoPlayoutIsAvailable(bool* available) const override {
     *available = false;
     return 0;
   }
-  int32_t StereoPlayout(bool* enabled) const override { return 0; }
+  int32_t StereoPlayout(bool* /* enabled */) const override { return 0; }
   int32_t StereoRecordingIsAvailable(bool* available) const override {
     *available = false;
     return 0;
   }
-  int32_t StereoRecording(bool* enabled) const override { return 0; }
+  int32_t StereoRecording(bool* /* enabled */) const override { return 0; }
   int32_t PlayoutDelay(uint16_t* delayMS) const override {
     *delayMS = 0;
     return 0;
   }
   bool BuiltInAECIsAvailable() const override { return false; }
-  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
+  int32_t EnableBuiltInAEC(bool /* enable */) override { return -1; }
   bool BuiltInAGCIsAvailable() const override { return false; }
-  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
+  int32_t EnableBuiltInAGC(bool /* enable */) override { return -1; }
   bool BuiltInNSIsAvailable() const override { return false; }
   int32_t EnableBuiltInNS(bool enable) override { return -1; }
 
diff --git a/modules/audio_device/include/test_audio_device_unittest.cc b/modules/audio_device/include/test_audio_device_unittest.cc
index a0eb240c5b..94592fdeb0 100644
--- a/modules/audio_device/include/test_audio_device_unittest.cc
+++ b/modules/audio_device/include/test_audio_device_unittest.cc
@@ -356,17 +356,17 @@ class TestAudioTransport : public AudioTransport {
   ~TestAudioTransport() override = default;
 
   int32_t RecordedDataIsAvailable(
-      const void* audioSamples,
+      const void* /* audioSamples */,
       size_t samples_per_channel,
       size_t bytes_per_sample,
       size_t number_of_channels,
       uint32_t samples_per_second,
-      uint32_t total_delay_ms,
-      int32_t clock_drift,
-      uint32_t current_mic_level,
-      bool key_pressed,
+      uint32_t /* total_delay_ms */,
+      int32_t /* clock_drift */,
+      uint32_t /* current_mic_level */,
+      bool /* key_pressed */,
       uint32_t& new_mic_level,
-      std::optional<int64_t> estimated_capture_time_ns) override {
+      std::optional<int64_t> /* estimated_capture_time_ns */) override {
     new_mic_level = 1;
 
     if (mode_ != Mode::kRecording) {
@@ -411,26 +411,26 @@ class TestAudioTransport : public AudioTransport {
     return 0;
   }
 
-  int32_t RecordedDataIsAvailable(const void* audio_samples,
-                                  size_t samples_per_channel,
-                                  size_t bytes_per_sample,
-                                  size_t number_of_channels,
-                                  uint32_t samples_per_second,
-                                  uint32_t total_delay_ms,
-                                  int32_t clockDrift,
-                                  uint32_t current_mic_level,
-                                  bool key_pressed,
-                                  uint32_t& new_mic_level) override {
+  int32_t RecordedDataIsAvailable(const void* /* audio_samples */,
+                                  size_t /* samples_per_channel */,
+                                  size_t /* bytes_per_sample */,
+                                  size_t /* number_of_channels */,
+                                  uint32_t /* samples_per_second */,
+                                  uint32_t /* total_delay_ms */,
+                                  int32_t /* clockDrift */,
+                                  uint32_t /* current_mic_level */,
+                                  bool /* key_pressed */,
+                                  uint32_t& /* new_mic_level */) override {
     RTC_CHECK(false) << "This methods should be never executed";
   }
 
-  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 {
+  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 {
     RTC_CHECK(false) << "This methods should be never executed";
   }
 
diff --git a/modules/audio_device/test_audio_device_impl.h b/modules/audio_device/test_audio_device_impl.h
index 87fd9a9630..0b12b8f4be 100644
--- a/modules/audio_device/test_audio_device_impl.h
+++ b/modules/audio_device/test_audio_device_impl.h
@@ -44,7 +44,7 @@ class TestAudioDevice : public AudioDeviceGeneric {
 
   // Retrieve the currently utilized audio layer
   int32_t ActiveAudioLayer(
-      AudioDeviceModule::AudioLayer& audioLayer) const override {
+      AudioDeviceModule::AudioLayer& /* audioLayer */) const override {
     return 0;
   }
 
@@ -56,26 +56,26 @@ class TestAudioDevice : public AudioDeviceGeneric {
   // Device enumeration
   int16_t PlayoutDevices() override { return 0; }
   int16_t RecordingDevices() override { return 0; }
-  int32_t PlayoutDeviceName(uint16_t index,
-                            char name[kAdmMaxDeviceNameSize],
-                            char guid[kAdmMaxGuidSize]) override {
+  int32_t PlayoutDeviceName(uint16_t /* index */,
+                            char /* name */[kAdmMaxDeviceNameSize],
+                            char /* guid */[kAdmMaxGuidSize]) override {
     return 0;
   }
-  int32_t RecordingDeviceName(uint16_t index,
-                              char name[kAdmMaxDeviceNameSize],
-                              char guid[kAdmMaxGuidSize]) override {
+  int32_t RecordingDeviceName(uint16_t /* index */,
+                              char /* name */[kAdmMaxDeviceNameSize],
+                              char /* guid */[kAdmMaxGuidSize]) override {
     return 0;
   }
 
   // Device selection
-  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
+  int32_t SetPlayoutDevice(uint16_t /* index */) override { return 0; }
   int32_t SetPlayoutDevice(
-      AudioDeviceModule::WindowsDeviceType device) override {
+      AudioDeviceModule::WindowsDeviceType /* device */) override {
     return 0;
   }
-  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
+  int32_t SetRecordingDevice(uint16_t /* index */) override { return 0; }
   int32_t SetRecordingDevice(
-      AudioDeviceModule::WindowsDeviceType device) override {
+      AudioDeviceModule::WindowsDeviceType /* device */) override {
     return 0;
   }
 
@@ -102,42 +102,54 @@ class TestAudioDevice : public AudioDeviceGeneric {
   bool MicrophoneIsInitialized() const override { return true; }
 
   // Speaker volume controls
-  int32_t SpeakerVolumeIsAvailable(bool& available) override { return 0; }
-  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
-  int32_t SpeakerVolume(uint32_t& volume) const override { return 0; }
-  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override { return 0; }
-  int32_t MinSpeakerVolume(uint32_t& minVolume) const override { return 0; }
+  int32_t SpeakerVolumeIsAvailable(bool& /* available */) override { return 0; }
+  int32_t SetSpeakerVolume(uint32_t /* volume */) override { return 0; }
+  int32_t SpeakerVolume(uint32_t& /* volume */) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t& /* maxVolume */) const override {
+    return 0;
+  }
+  int32_t MinSpeakerVolume(uint32_t& /* minVolume */) const override {
+    return 0;
+  }
 
   // Microphone volume controls
-  int32_t MicrophoneVolumeIsAvailable(bool& available) override { return 0; }
-  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
-  int32_t MicrophoneVolume(uint32_t& volume) const override { return 0; }
-  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override { return 0; }
-  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override { return 0; }
+  int32_t MicrophoneVolumeIsAvailable(bool& /* available */) override {
+    return 0;
+  }
+  int32_t SetMicrophoneVolume(uint32_t /* volume */) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t& /* volume */) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t& /* maxVolume */) const override {
+    return 0;
+  }
+  int32_t MinMicrophoneVolume(uint32_t& /* minVolume */) const override {
+    return 0;
+  }
 
   // Speaker mute control
-  int32_t SpeakerMuteIsAvailable(bool& available) override { return 0; }
-  int32_t SetSpeakerMute(bool enable) override { return 0; }
-  int32_t SpeakerMute(bool& enabled) const override { return 0; }
+  int32_t SpeakerMuteIsAvailable(bool& /* available */) override { return 0; }
+  int32_t SetSpeakerMute(bool /* enable */) override { return 0; }
+  int32_t SpeakerMute(bool& /* enabled */) const override { return 0; }
 
   // Microphone mute control
-  int32_t MicrophoneMuteIsAvailable(bool& available) override { return 0; }
-  int32_t SetMicrophoneMute(bool enable) override { return 0; }
-  int32_t MicrophoneMute(bool& enabled) const override { return 0; }
+  int32_t MicrophoneMuteIsAvailable(bool& /* available */) override {
+    return 0;
+  }
+  int32_t SetMicrophoneMute(bool /* enable */) override { return 0; }
+  int32_t MicrophoneMute(bool& /* enabled */) const override { return 0; }
 
   // Stereo support
   int32_t StereoPlayoutIsAvailable(bool& available) override {
     available = false;
     return 0;
   }
-  int32_t SetStereoPlayout(bool enable) override { return 0; }
-  int32_t StereoPlayout(bool& enabled) const override { return 0; }
+  int32_t SetStereoPlayout(bool /* enable */) override { return 0; }
+  int32_t StereoPlayout(bool& /* enabled */) const override { return 0; }
   int32_t StereoRecordingIsAvailable(bool& available) override {
     available = false;
     return 0;
   }
-  int32_t SetStereoRecording(bool enable) override { return 0; }
-  int32_t StereoRecording(bool& enabled) const override { return 0; }
+  int32_t SetStereoRecording(bool /* enable */) override { return 0; }
+  int32_t StereoRecording(bool& /* enabled */) const override { return 0; }
 
   // Delay information and control
   int32_t PlayoutDelay(uint16_t& delayMS) const override {
@@ -151,9 +163,9 @@ class TestAudioDevice : public AudioDeviceGeneric {
   bool BuiltInNSIsAvailable() const override { return false; }
 
   // Windows Core Audio and Android only.
-  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
-  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
-  int32_t EnableBuiltInNS(bool enable) override { return -1; }
+  int32_t EnableBuiltInAEC(bool /* enable */) override { return -1; }
+  int32_t EnableBuiltInAGC(bool /* enable */) override { return -1; }
+  int32_t EnableBuiltInNS(bool /* enable */) override { return -1; }
 
   // Play underrun count.
   int32_t GetPlayoutUnderrunCount() const override { return -1; }
diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc
index 00e2f7a9ee..2b8a6209d0 100644
--- a/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -121,7 +121,7 @@ class CustomRateCalculator : public OutputRateCalculator {
  public:
   explicit CustomRateCalculator(int rate) : rate_(rate) {}
   int CalculateOutputRateFromRange(
-      rtc::ArrayView<const int> preferred_rates) override {
+      rtc::ArrayView<const int> /* preferred_rates */) override {
     return rate_;
   }
 
@@ -483,7 +483,7 @@ class HighOutputRateCalculator : public OutputRateCalculator {
  public:
   static const int kDefaultFrequency = 76000;
   int CalculateOutputRateFromRange(
-      rtc::ArrayView<const int> preferred_sample_rates) override {
+      rtc::ArrayView<const int> /* preferred_sample_rates */) override {
     return kDefaultFrequency;
   }
   ~HighOutputRateCalculator() override {}
diff --git a/modules/audio_mixer/frame_combiner.cc b/modules/audio_mixer/frame_combiner.cc
index dfe9511f5a..eb269f61fb 100644
--- a/modules/audio_mixer/frame_combiner.cc
+++ b/modules/audio_mixer/frame_combiner.cc
@@ -39,7 +39,7 @@ namespace {
 void SetAudioFrameFields(rtc::ArrayView<const AudioFrame* const> mix_list,
                          size_t number_of_channels,
                          int sample_rate,
-                         size_t number_of_streams,
+                         size_t /* number_of_streams */,
                          AudioFrame* audio_frame_for_mixing) {
   const size_t samples_per_channel =
       SampleRateToDefaultChannelSize(sample_rate);
diff --git a/modules/audio_processing/aec3/block_processor.cc b/modules/audio_processing/aec3/block_processor.cc
index ad3702e165..010a76cd90 100644
--- a/modules/audio_processing/aec3/block_processor.cc
+++ b/modules/audio_processing/aec3/block_processor.cc
@@ -84,8 +84,8 @@ std::atomic<int> BlockProcessorImpl::instance_count_(0);
 BlockProcessorImpl::BlockProcessorImpl(
     const EchoCanceller3Config& config,
     int sample_rate_hz,
-    size_t num_render_channels,
-    size_t num_capture_channels,
+    size_t /* num_render_channels */,
+    size_t /* num_capture_channels */,
     std::unique_ptr<RenderDelayBuffer> render_buffer,
     std::unique_ptr<RenderDelayController> delay_controller,
     std::unique_ptr<EchoRemover> echo_remover)
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 9cc11dfbc4..5db489e42a 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -107,7 +107,7 @@ bool VerifyOutputFrameBitexactness(rtc::ArrayView<const float> reference,
 // EchoCanceller3 output.
 class CaptureTransportVerificationProcessor : public BlockProcessor {
  public:
-  explicit CaptureTransportVerificationProcessor(size_t num_bands) {}
+  explicit CaptureTransportVerificationProcessor(size_t /* num_bands */) {}
 
   CaptureTransportVerificationProcessor() = delete;
   CaptureTransportVerificationProcessor(
@@ -117,27 +117,27 @@ class CaptureTransportVerificationProcessor : public BlockProcessor {
 
   ~CaptureTransportVerificationProcessor() override = default;
 
-  void ProcessCapture(bool level_change,
-                      bool saturated_microphone_signal,
-                      Block* linear_output,
-                      Block* capture_block) override {}
+  void ProcessCapture(bool /* level_change */,
+                      bool /* saturated_microphone_signal */,
+                      Block* /* linear_output */,
+                      Block* /* capture_block */) override {}
 
-  void BufferRender(const Block& block) override {}
+  void BufferRender(const Block& /* block */) override {}
 
-  void UpdateEchoLeakageStatus(bool leakage_detected) override {}
+  void UpdateEchoLeakageStatus(bool /* leakage_detected */) override {}
 
-  void GetMetrics(EchoControl::Metrics* metrics) const override {}
+  void GetMetrics(EchoControl::Metrics* /* metrics */) const override {}
 
-  void SetAudioBufferDelay(int delay_ms) override {}
+  void SetAudioBufferDelay(int /* delay_ms */) override {}
 
-  void SetCaptureOutputUsage(bool capture_output_used) {}
+  void SetCaptureOutputUsage(bool /* capture_output_used */) {}
 };
 
 // Class for testing that the render data is properly received by the block
 // processor.
 class RenderTransportVerificationProcessor : public BlockProcessor {
  public:
-  explicit RenderTransportVerificationProcessor(size_t num_bands) {}
+  explicit RenderTransportVerificationProcessor(size_t /* num_bands */) {}
 
   RenderTransportVerificationProcessor() = delete;
   RenderTransportVerificationProcessor(
@@ -147,9 +147,9 @@ class RenderTransportVerificationProcessor : public BlockProcessor {
 
   ~RenderTransportVerificationProcessor() override = default;
 
-  void ProcessCapture(bool level_change,
-                      bool saturated_microphone_signal,
-                      Block* linear_output,
+  void ProcessCapture(bool /* level_change */,
+                      bool /* saturated_microphone_signal */,
+                      Block* /* linear_output */,
                       Block* capture_block) override {
     Block render_block = received_render_blocks_.front();
     received_render_blocks_.pop_front();
@@ -160,9 +160,9 @@ class RenderTransportVerificationProcessor : public BlockProcessor {
     received_render_blocks_.push_back(block);
   }
 
-  void UpdateEchoLeakageStatus(bool leakage_detected) override {}
+  void UpdateEchoLeakageStatus(bool /* leakage_detected */) override {}
 
-  void GetMetrics(EchoControl::Metrics* metrics) const override {}
+  void GetMetrics(EchoControl::Metrics* /* metrics */) const override {}
 
   void SetAudioBufferDelay(int delay_ms) override {}
 
diff --git a/modules/audio_processing/aec3/echo_remover_metrics.cc b/modules/audio_processing/aec3/echo_remover_metrics.cc
index c3fc80773a..1e3a70ad38 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics.cc
@@ -53,8 +53,8 @@ void EchoRemoverMetrics::ResetMetrics() {
 
 void EchoRemoverMetrics::Update(
     const AecState& aec_state,
-    const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
-    const std::array<float, kFftLengthBy2Plus1>& suppressor_gain) {
+    const std::array<float, kFftLengthBy2Plus1>& /* comfort_noise_spectrum */,
+    const std::array<float, kFftLengthBy2Plus1>& /* suppressor_gain */) {
   metrics_reported_ = false;
   if (++block_counter_ <= kMetricsCollectionBlocks) {
     erl_time_domain_.UpdateInstant(aec_state.ErlTimeDomain());
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index 59a3b46872..8800f4a1bd 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -780,7 +780,7 @@ void MatchedFilter::Update(const DownsampledRenderBuffer& render_buffer,
   }
 }
 
-void MatchedFilter::LogFilterProperties(int sample_rate_hz,
+void MatchedFilter::LogFilterProperties(int /* sample_rate_hz */,
                                         size_t shift,
                                         size_t downsampling_factor) const {
   size_t alignment_shift = 0;
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index df594ea57a..b02e516a04 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -113,7 +113,7 @@ void RenderDelayControllerImpl::LogRenderCall() {}
 
 std::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
     const DownsampledRenderBuffer& render_buffer,
-    size_t render_delay_buffer_delay,
+    size_t /* render_delay_buffer_delay */,
     const Block& capture) {
   ++capture_call_counter_;
 
diff --git a/modules/audio_processing/aec3/subband_nearend_detector.cc b/modules/audio_processing/aec3/subband_nearend_detector.cc
index 2aa400c3af..c8998a5a22 100644
--- a/modules/audio_processing/aec3/subband_nearend_detector.cc
+++ b/modules/audio_processing/aec3/subband_nearend_detector.cc
@@ -30,10 +30,10 @@ void SubbandNearendDetector::Update(
     rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
         nearend_spectrum,
     rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
-        residual_echo_spectrum,
+    /* residual_echo_spectrum */,
     rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
         comfort_noise_spectrum,
-    bool initial_state) {
+    bool /* initial_state */) {
   nearend_state_ = false;
   for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
     const std::array<float, kFftLengthBy2Plus1>& noise =
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index 26797fc0b1..9b196e3163 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -323,7 +323,7 @@ void SuppressionGain::LowerBandGain(
 
 SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
                                  Aec3Optimization optimization,
-                                 int sample_rate_hz,
+                                 int /* sample_rate_hz */,
                                  size_t num_capture_channels)
     : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
       optimization_(optimization),
diff --git a/modules/audio_processing/aec3/transparent_mode.cc b/modules/audio_processing/aec3/transparent_mode.cc
index 4d6937ff96..a5929c41d9 100644
--- a/modules/audio_processing/aec3/transparent_mode.cc
+++ b/modules/audio_processing/aec3/transparent_mode.cc
@@ -45,13 +45,13 @@ class TransparentModeImpl : public TransparentMode {
     prob_transparent_state_ = kInitialTransparentStateProbability;
   }
 
-  void Update(int filter_delay_blocks,
-              bool any_filter_consistent,
-              bool any_filter_converged,
+  void Update(int /* filter_delay_blocks */,
+              bool /* any_filter_consistent */,
+              bool /* any_filter_converged */,
               bool any_coarse_filter_converged,
-              bool all_filters_diverged,
+              bool /* all_filters_diverged */,
               bool active_render,
-              bool saturated_capture) override {
+              bool /* saturated_capture */) override {
     // The classifier is implemented as a Hidden Markov Model (HMM) with two
     // hidden states: "normal" and "transparent". The estimated probabilities of
     // the two states are updated by observing filter convergence during active
@@ -145,7 +145,7 @@ class LegacyTransparentModeImpl : public TransparentMode {
   void Update(int filter_delay_blocks,
               bool any_filter_consistent,
               bool any_filter_converged,
-              bool any_coarse_filter_converged,
+              bool /* any_coarse_filter_converged */,
               bool all_filters_diverged,
               bool active_render,
               bool saturated_capture) override {
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 18d84d8a18..763dd9ace6 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -160,7 +160,7 @@ int GetSpeechLevelErrorDb(float speech_level_dbfs, float speech_probability) {
 
 }  // namespace
 
-MonoAgc::MonoAgc(ApmDataDumper* data_dumper,
+MonoAgc::MonoAgc(ApmDataDumper* /* data_dumper */,
                  int clipped_level_min,
                  bool disable_digital_adaptive,
                  int min_mic_level)
diff --git a/modules/audio_processing/agc/legacy/digital_agc.cc b/modules/audio_processing/agc/legacy/digital_agc.cc
index 4cd86acba8..5bf7aaff57 100644
--- a/modules/audio_processing/agc/legacy/digital_agc.cc
+++ b/modules/audio_processing/agc/legacy/digital_agc.cc
@@ -271,7 +271,7 @@ int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
 // Gains is an 11 element long array (one value per ms, incl start & end).
 int32_t WebRtcAgc_ComputeDigitalGains(DigitalAgc* stt,
                                       const int16_t* const* in_near,
-                                      size_t num_bands,
+                                      size_t /* num_bands */,
                                       uint32_t FS,
                                       int16_t lowlevelSignal,
                                       int32_t gains[11]) {
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index e8c912518d..04b2384930 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -182,11 +182,11 @@ void ComputeAutoCorrelation(
 // Searches the strongest pitch period at 24 kHz and returns its inverted lag at
 // 48 kHz.
 int ComputePitchPeriod48kHz(
-    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kBufSize24kHz> /* pitch_buffer */,
     rtc::ArrayView<const int> inverted_lags,
     rtc::ArrayView<const float, kInitialNumLags24kHz> auto_correlation,
     rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
-    const VectorMath& vector_math) {
+    const VectorMath& /* vector_math */) {
   static_assert(kMaxPitch24kHz > kInitialNumLags24kHz, "");
   static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
   int best_inverted_lag = 0;     // Pitch period.
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index c48d444664..684554faae 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -42,7 +42,7 @@ AudioBuffer::AudioBuffer(size_t input_rate,
                          size_t buffer_rate,
                          size_t buffer_num_channels,
                          size_t output_rate,
-                         size_t output_num_channels)
+                         size_t /* output_num_channels */)
     : input_num_frames_(static_cast<int>(input_rate) / 100),
       input_num_channels_(input_num_channels),
       buffer_num_frames_(static_cast<int>(buffer_rate) / 100),
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index a1cba51a11..ed80ba4328 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1579,8 +1579,8 @@ int AudioProcessingImpl::ProcessReverseStream(const float* const* src,
 
 int AudioProcessingImpl::AnalyzeReverseStreamLocked(
     const float* const* src,
-    const StreamConfig& input_config,
-    const StreamConfig& output_config) {
+    const StreamConfig& /* input_config */,
+    const StreamConfig& /* output_config */) {
   if (aec_dump_) {
     const size_t channel_size =
         formats_.api_format.reverse_input_stream().num_frames();
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index ecdc055768..19c6dccf43 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -130,7 +130,7 @@ class AudioProcessingImpl : public AudioProcessing {
   size_t num_reverse_channels() const override;
   int stream_delay_ms() const override;
 
-  AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
+  AudioProcessingStats GetStatistics(bool /* has_remote_tracks */) override {
     return GetStatistics();
   }
   AudioProcessingStats GetStatistics() override {
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 1c3c03c013..cebc60aeaa 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -63,9 +63,9 @@ class MockEchoControlFactory : public EchoControlFactory {
   MockEchoControlFactory() : next_mock_(std::make_unique<MockEchoControl>()) {}
   // Returns a pointer to the next MockEchoControl that this factory creates.
   MockEchoControl* GetNext() const { return next_mock_.get(); }
-  std::unique_ptr<EchoControl> Create(int sample_rate_hz,
-                                      int num_render_channels,
-                                      int num_capture_channels) override {
+  std::unique_ptr<EchoControl> Create(int /* sample_rate_hz */,
+                                      int /* num_render_channels */,
+                                      int /* num_capture_channels */) override {
     std::unique_ptr<EchoControl> mock = std::move(next_mock_);
     next_mock_ = std::make_unique<MockEchoControl>();
     return mock;
@@ -88,12 +88,12 @@ class TestEchoDetector : public EchoDetector {
     last_render_audio_first_sample_ = render_audio[0];
     analyze_render_audio_called_ = true;
   }
-  void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
-  }
-  void Initialize(int capture_sample_rate_hz,
-                  int num_capture_channels,
-                  int render_sample_rate_hz,
-                  int num_render_channels) override {}
+  void AnalyzeCaptureAudio(
+      rtc::ArrayView<const float> /* capture_audio */) override {}
+  void Initialize(int /* capture_sample_rate_hz */,
+                  int /* num_capture_channels */,
+                  int /* render_sample_rate_hz */,
+                  int /* num_render_channels */) override {}
   EchoDetector::Metrics GetMetrics() const override { return {}; }
   // Returns true if AnalyzeRenderAudio() has been called at least once.
   bool analyze_render_audio_called() const {
@@ -116,7 +116,7 @@ class TestRenderPreProcessor : public CustomProcessing {
  public:
   TestRenderPreProcessor() = default;
   ~TestRenderPreProcessor() = default;
-  void Initialize(int sample_rate_hz, int num_channels) override {}
+  void Initialize(int /* sample_rate_hz */, int /* num_channels */) override {}
   void Process(AudioBuffer* audio) override {
     for (size_t k = 0; k < audio->num_channels(); ++k) {
       rtc::ArrayView<float> channel_view(audio->channels()[k],
@@ -126,7 +126,8 @@ class TestRenderPreProcessor : public CustomProcessing {
     }
   }
   std::string ToString() const override { return "TestRenderPreProcessor"; }
-  void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
+  void SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting /* setting */) override {}
   // Modifies a sample. This member is used in Process() to modify a frame and
   // it is publicly visible to enable tests.
   static constexpr float ProcessSample(float x) { return 2.f * x; }
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index cfe25107ca..db4a819805 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -592,8 +592,8 @@ bool ApmTest::ReadFrame(FILE* file, Int16FrameData* frame) {
 
 // If the end of the file has been reached, rewind it and attempt to read the
 // frame again.
-void ApmTest::ReadFrameWithRewind(FILE* file,
-                                  Int16FrameData* frame,
+void ApmTest::ReadFrameWithRewind(FILE* /* file */,
+                                  Int16FrameData* /* frame */,
                                   ChannelBuffer<float>* cb) {
   if (!ReadFrame(near_file_, &frame_, cb)) {
     rewind(near_file_);
@@ -2619,7 +2619,7 @@ TEST(ApmConfiguration, PreProcessingReceivesRuntimeSettings) {
 
 class MyEchoControlFactory : public EchoControlFactory {
  public:
-  std::unique_ptr<EchoControl> Create(int sample_rate_hz) {
+  std::unique_ptr<EchoControl> Create(int /* sample_rate_hz */) {
     auto ec = new test::MockEchoControl();
     EXPECT_CALL(*ec, AnalyzeRender(::testing::_)).Times(1);
     EXPECT_CALL(*ec, AnalyzeCapture(::testing::_)).Times(2);
@@ -2629,8 +2629,8 @@ class MyEchoControlFactory : public EchoControlFactory {
   }
 
   std::unique_ptr<EchoControl> Create(int sample_rate_hz,
-                                      int num_render_channels,
-                                      int num_capture_channels) {
+                                      int /* num_render_channels */,
+                                      int /* num_capture_channels */) {
     return Create(sample_rate_hz);
   }
 };
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index 283f7392bf..f92d4ba0d8 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -54,7 +54,7 @@ std::string FormFileName(absl::string_view output_dir,
 ApmDataDumper::ApmDataDumper(int instance_index)
     : instance_index_(instance_index) {}
 #else
-ApmDataDumper::ApmDataDumper(int instance_index) {}
+ApmDataDumper::ApmDataDumper(int /* instance_index */) {}
 #endif
 
 ApmDataDumper::~ApmDataDumper() = default;
diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index 5186854dd5..09d24bd034 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -61,7 +61,7 @@ class ApmDataDumper {
   ~ApmDataDumper();
 
   // Activates or deactivate the dumping functionality.
-  static void SetActivated(bool activated) {
+  static void SetActivated([[maybe_unused]] bool activated) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     recording_activated_ = activated;
 #endif
@@ -82,14 +82,15 @@ class ApmDataDumper {
   // Specifies what dump set to use. All dump commands with a different dump set
   // than the one specified will be discarded. If not specificed, all dump sets
   // will be used.
-  static void SetDumpSetToUse(int dump_set_to_use) {
+  static void SetDumpSetToUse([[maybe_unused]] int dump_set_to_use) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     dump_set_to_use_ = dump_set_to_use;
 #endif
   }
 
   // Set an optional output directory.
-  static void SetOutputDirectory(absl::string_view output_dir) {
+  static void SetOutputDirectory(
+      [[maybe_unused]] absl::string_view output_dir) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
     rtc::strcpyn(output_dir_, kOutputDirMaxLength, output_dir);
@@ -106,9 +107,9 @@ class ApmDataDumper {
 
   // Methods for performing dumping of data of various types into
   // various formats.
-  void DumpRaw(absl::string_view name,
-               double v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] double v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -120,10 +121,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const double* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const double* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -135,9 +136,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const double> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const double> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -148,9 +149,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               float v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] float v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -162,10 +163,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const float* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const float* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -177,9 +178,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const float> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const float> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -190,7 +191,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name, bool v, int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] bool v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -201,10 +204,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const bool* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const bool* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -219,9 +222,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const bool> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const bool> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -232,9 +235,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               int16_t v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] int16_t v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -246,10 +249,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const int16_t* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const int16_t* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -261,9 +264,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const int16_t> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const int16_t> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -274,9 +277,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               int32_t v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] int32_t v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -288,10 +291,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const int32_t* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const int32_t* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -303,9 +306,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -317,10 +320,10 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               size_t v_length,
-               const size_t* v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const size_t* v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -332,9 +335,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const int32_t> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const int32_t> v,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -345,9 +348,9 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpRaw(absl::string_view name,
-               rtc::ArrayView<const size_t> v,
-               int dump_set = kDefaultDumpSet) {
+  void DumpRaw(absl::string_view /* name */,
+               rtc::ArrayView<const size_t> /* v */,
+               int /* dump_set */ = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -356,12 +359,12 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpWav(absl::string_view name,
-               size_t v_length,
-               const float* v,
-               int sample_rate_hz,
-               int num_channels,
-               int dump_set = kDefaultDumpSet) {
+  void DumpWav([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] size_t v_length,
+               [[maybe_unused]] const float* v,
+               [[maybe_unused]] int sample_rate_hz,
+               [[maybe_unused]] int num_channels,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
@@ -374,11 +377,11 @@ class ApmDataDumper {
 #endif
   }
 
-  void DumpWav(absl::string_view name,
-               rtc::ArrayView<const float> v,
-               int sample_rate_hz,
-               int num_channels,
-               int dump_set = kDefaultDumpSet) {
+  void DumpWav([[maybe_unused]] absl::string_view name,
+               [[maybe_unused]] rtc::ArrayView<const float> v,
+               [[maybe_unused]] int sample_rate_hz,
+               [[maybe_unused]] int num_channels,
+               [[maybe_unused]] int dump_set = kDefaultDumpSet) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
     if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
       return;
diff --git a/modules/audio_processing/ns/noise_suppressor_unittest.cc b/modules/audio_processing/ns/noise_suppressor_unittest.cc
index 28ea63ae40..8927b6b67f 100644
--- a/modules/audio_processing/ns/noise_suppressor_unittest.cc
+++ b/modules/audio_processing/ns/noise_suppressor_unittest.cc
@@ -48,7 +48,7 @@ void PopulateInputFrameWithIdenticalChannels(size_t num_channels,
 
 void VerifyIdenticalChannels(size_t num_channels,
                              size_t num_bands,
-                             size_t frame_index,
+                             size_t /* frame_index */,
                              const AudioBuffer& audio) {
   EXPECT_GT(num_channels, 1u);
   for (size_t ch = 1; ch < num_channels; ++ch) {
diff --git a/modules/audio_processing/splitting_filter.cc b/modules/audio_processing/splitting_filter.cc
index d47090bc03..0b7399bf19 100644
--- a/modules/audio_processing/splitting_filter.cc
+++ b/modules/audio_processing/splitting_filter.cc
@@ -27,7 +27,7 @@ constexpr size_t kTwoBandFilterSamplesPerFrame = 320;
 
 SplittingFilter::SplittingFilter(size_t num_channels,
                                  size_t num_bands,
-                                 size_t num_frames)
+                                 size_t /* num_frames */)
     : num_bands_(num_bands),
       two_bands_states_(num_bands_ == 2 ? num_channels : 0),
       three_band_filter_banks_(num_bands_ == 3 ? num_channels : 0) {
diff --git a/modules/audio_processing/test/fake_recording_device.cc b/modules/audio_processing/test/fake_recording_device.cc
index 476ab8a23a..41cd3b8c42 100644
--- a/modules/audio_processing/test/fake_recording_device.cc
+++ b/modules/audio_processing/test/fake_recording_device.cc
@@ -57,8 +57,8 @@ class FakeRecordingDeviceIdentity final : public FakeRecordingDeviceWorker {
   explicit FakeRecordingDeviceIdentity(const int initial_mic_level)
       : FakeRecordingDeviceWorker(initial_mic_level) {}
   ~FakeRecordingDeviceIdentity() override = default;
-  void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override {}
-  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {}
+  void ModifyBufferInt16(rtc::ArrayView<int16_t> /* buffer */) override {}
+  void ModifyBufferFloat(ChannelBuffer<float>* /* buffer */) override {}
 };
 
 // Linear fake recording device. The gain curve is a linear function mapping the
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
index 4310970930..d330cad6e7 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
@@ -35,7 +35,7 @@ AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
 AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
 
 AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
-    const FieldTrialsView* key_value_config,
+    const FieldTrialsView* /* key_value_config */,
     std::unique_ptr<BitrateEstimator> bitrate_estimator)
     : in_alr_(false), bitrate_estimator_(std::move(bitrate_estimator)) {}
 
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
index e2ab10484e..9769792de3 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
@@ -212,9 +212,9 @@ DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
 DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
     std::optional<DataRate> acked_bitrate,
     std::optional<DataRate> probe_bitrate,
-    std::optional<NetworkStateEstimate> state_estimate,
+    std::optional<NetworkStateEstimate> /* state_estimate */,
     bool recovered_from_overuse,
-    bool in_alr,
+    bool /* in_alr */,
     Timestamp at_time) {
   Result result;
 
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
index 433d3cbba5..c93bf29c51 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -232,7 +232,7 @@ void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
 // Returns true if an over-use was seen, false otherwise.
 // The StreamGenerator::updated() should be used to check for any changes in
 // target bitrate after the call to this function.
-bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
+bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t /* ssrc */,
                                                 uint32_t bitrate_bps) {
   stream_generator_->SetBitrateBps(bitrate_bps);
   std::vector<PacketResult> packets;
@@ -419,7 +419,7 @@ void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
 
 void DelayBasedBweTest::CapacityDropTestHelper(
     int number_of_streams,
-    bool wrap_time_stamp,
+    bool /* wrap_time_stamp */,
     uint32_t expected_bitrate_drop_delta,
     int64_t receiver_clock_offset_change_ms) {
   const int kFramerate = 30;
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
index 9b005118e9..11f5f588ff 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -300,7 +300,7 @@ NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
 }
 
 NetworkControlUpdate GoogCcNetworkController::OnReceivedPacket(
-    ReceivedPacket received_packet) {
+    ReceivedPacket /* received_packet */) {
   return NetworkControlUpdate();
 }
 
diff --git a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
index 4a32609030..db5df37156 100644
--- a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
+++ b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -372,8 +372,8 @@ void SendSideBandwidthEstimation::SetAcknowledgedRate(
 
 void SendSideBandwidthEstimation::UpdateLossBasedEstimator(
     const TransportPacketsFeedback& report,
-    BandwidthUsage delay_detector_state,
-    std::optional<DataRate> probe_bitrate,
+    BandwidthUsage /* delay_detector_state */,
+    std::optional<DataRate> /* probe_bitrate */,
     bool in_alr) {
   if (LossBasedBandwidthEstimatorV1Enabled()) {
     loss_based_bandwidth_estimator_v1_.UpdateLossStatistics(
diff --git a/modules/congestion_controller/goog_cc/trendline_estimator.cc b/modules/congestion_controller/goog_cc/trendline_estimator.cc
index 4fb11b3e84..66730a2f96 100644
--- a/modules/congestion_controller/goog_cc/trendline_estimator.cc
+++ b/modules/congestion_controller/goog_cc/trendline_estimator.cc
@@ -199,9 +199,9 @@ TrendlineEstimator::~TrendlineEstimator() {}
 
 void TrendlineEstimator::UpdateTrendline(double recv_delta_ms,
                                          double send_delta_ms,
-                                         int64_t send_time_ms,
+                                         int64_t /* send_time_ms */,
                                          int64_t arrival_time_ms,
-                                         size_t packet_size) {
+                                         size_t /* packet_size */) {
   const double delta_ms = recv_delta_ms - send_delta_ms;
   ++num_of_deltas_;
   num_of_deltas_ = std::min(num_of_deltas_, kDeltaCounterMax);
diff --git a/modules/congestion_controller/pcc/pcc_network_controller.cc b/modules/congestion_controller/pcc/pcc_network_controller.cc
index 1705afb537..76f91bcb69 100644
--- a/modules/congestion_controller/pcc/pcc_network_controller.cc
+++ b/modules/congestion_controller/pcc/pcc_network_controller.cc
@@ -339,12 +339,12 @@ void PccNetworkController::UpdateSendingRateAndMode() {
 }
 
 NetworkControlUpdate PccNetworkController::OnNetworkAvailability(
-    NetworkAvailability msg) {
+    NetworkAvailability /* msg */) {
   return NetworkControlUpdate();
 }
 
 NetworkControlUpdate PccNetworkController::OnNetworkRouteChange(
-    NetworkRouteChange msg) {
+    NetworkRouteChange /* msg */) {
   return NetworkControlUpdate();
 }
 
@@ -354,7 +354,7 @@ NetworkControlUpdate PccNetworkController::OnProcessInterval(
 }
 
 NetworkControlUpdate PccNetworkController::OnTargetRateConstraints(
-    TargetRateConstraints msg) {
+    TargetRateConstraints /* msg */) {
   return NetworkControlUpdate();
 }
 
@@ -373,17 +373,18 @@ NetworkControlUpdate PccNetworkController::OnTransportLossReport(
   return NetworkControlUpdate();
 }
 
-NetworkControlUpdate PccNetworkController::OnStreamsConfig(StreamsConfig msg) {
+NetworkControlUpdate PccNetworkController::OnStreamsConfig(
+    StreamsConfig /* msg */) {
   return NetworkControlUpdate();
 }
 
 NetworkControlUpdate PccNetworkController::OnReceivedPacket(
-    ReceivedPacket msg) {
+    ReceivedPacket /* msg */) {
   return NetworkControlUpdate();
 }
 
 NetworkControlUpdate PccNetworkController::OnNetworkStateEstimate(
-    NetworkStateEstimate msg) {
+    NetworkStateEstimate /* msg */) {
   return NetworkControlUpdate();
 }
 
diff --git a/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc b/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
index 179e002bc5..45673722fe 100644
--- a/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
+++ b/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
@@ -121,7 +121,9 @@ class FakeScreenCapturer : public DesktopCapturer {
     next_frame_ = std::move(next_frame);
   }
 
-  bool IsOccluded(const DesktopVector& pos) override { return is_occluded_; }
+  bool IsOccluded(const DesktopVector& /* pos */) override {
+    return is_occluded_;
+  }
 
   void set_is_occluded(bool value) { is_occluded_ = value; }
 
@@ -147,7 +149,9 @@ class FakeMouseMonitor : public MouseCursorMonitor {
     hotspot_ = hotspot;
   }
 
-  void Init(Callback* callback, Mode mode) override { callback_ = callback; }
+  void Init(Callback* callback, Mode /* mode */) override {
+    callback_ = callback;
+  }
 
   void Capture() override {
     if (changed_) {
@@ -209,7 +213,7 @@ class DesktopAndCursorComposerTest : public ::testing::Test,
   }
 
   // DesktopCapturer::Callback interface
-  void OnCaptureResult(DesktopCapturer::Result result,
+  void OnCaptureResult(DesktopCapturer::Result /* result */,
                        std::unique_ptr<DesktopFrame> frame) override {
     frame_ = std::move(frame);
   }
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index a52a76c262..7fd0fc31d8 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -45,15 +45,15 @@ DesktopCapturer::GetDelegatedSourceListController() {
 }
 
 void DesktopCapturer::SetSharedMemoryFactory(
-    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {}
+    std::unique_ptr<SharedMemoryFactory> /* shared_memory_factory */) {}
 
-void DesktopCapturer::SetExcludedWindow(WindowId window) {}
+void DesktopCapturer::SetExcludedWindow(WindowId /* window */) {}
 
-bool DesktopCapturer::GetSourceList(SourceList* sources) {
+bool DesktopCapturer::GetSourceList(SourceList* /* sources */) {
   return true;
 }
 
-bool DesktopCapturer::SelectSource(SourceId id) {
+bool DesktopCapturer::SelectSource(SourceId /* id */) {
   return false;
 }
 
@@ -61,7 +61,7 @@ bool DesktopCapturer::FocusOnSelectedSource() {
   return false;
 }
 
-bool DesktopCapturer::IsOccluded(const DesktopVector& pos) {
+bool DesktopCapturer::IsOccluded(const DesktopVector& /* pos */) {
   return false;
 }
 
@@ -109,7 +109,7 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
 
 // static
 std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
-    const DesktopCaptureOptions& options) {
+    [[maybe_unused]] const DesktopCaptureOptions& options) {
   std::unique_ptr<DesktopCapturer> capturer;
 
 #if defined(WEBRTC_USE_PIPEWIRE)
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 9c7ecc78f4..bb0742a410 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -109,7 +109,7 @@ class RTC_EXPORT DesktopCapturer {
   // supported by all capturers. This will only affect the frequency at which
   // new frames are available, not the frequency at which you are allowed to
   // capture the frames.
-  virtual void SetMaxFrameRate(uint32_t max_frame_rate) {}
+  virtual void SetMaxFrameRate(uint32_t /* max_frame_rate */) {}
 
   // Returns a valid pointer if the capturer requires the user to make a
   // selection from a source list provided by the capturer.
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.mm b/modules/desktop_capture/mac/screen_capturer_mac.mm
index e6746578ad..d4e9c111db 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.mm
+++ b/modules/desktop_capture/mac/screen_capturer_mac.mm
@@ -455,7 +455,7 @@ bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
     DesktopVector display_origin = config.pixel_bounds.top_left();
 
     CGDisplayStreamFrameAvailableHandler handler = ^(CGDisplayStreamFrameStatus status,
-                                                     uint64_t display_time,
+                                                     uint64_t /* display_time */,
                                                      IOSurfaceRef frame_surface,
                                                      CGDisplayStreamUpdateRef updateRef) {
       RTC_DCHECK(thread_checker_.IsCurrent());
diff --git a/modules/desktop_capture/mac/screen_capturer_sck.mm b/modules/desktop_capture/mac/screen_capturer_sck.mm
index fe44694d3a..9e2d9047e6 100644
--- a/modules/desktop_capture/mac/screen_capturer_sck.mm
+++ b/modules/desktop_capture/mac/screen_capturer_sck.mm
@@ -138,7 +138,7 @@ void ScreenCapturerSck::Start(DesktopCapturer::Callback* callback) {
   StartOrReconfigureCapturer();
 }
 
-void ScreenCapturerSck::SetMaxFrameRate(uint32_t max_frame_rate) {
+void ScreenCapturerSck::SetMaxFrameRate(uint32_t /* max_frame_rate */) {
   // TODO: crbug.com/327458809 - Implement this.
 }
 
@@ -345,7 +345,7 @@ void ScreenCapturerSck::StartOrReconfigureCapturer() {
   // inside the block is equivalent to `this->helper_` and would crash (UAF) if `this` is
   // deleted before the block is executed.
   SckHelper* local_helper = helper_;
-  auto handler = ^(SCShareableContent* content, NSError* error) {
+  auto handler = ^(SCShareableContent* content, NSError* /* error */) {
     [local_helper onShareableContentCreated:content];
   };
 
diff --git a/modules/desktop_capture/mouse_cursor_monitor.h b/modules/desktop_capture/mouse_cursor_monitor.h
index ad134df36d..31549cbc81 100644
--- a/modules/desktop_capture/mouse_cursor_monitor.h
+++ b/modules/desktop_capture/mouse_cursor_monitor.h
@@ -53,8 +53,8 @@ class MouseCursorMonitor {
     // Called in response to Capture(). `position` indicates cursor position
     // relative to the `window` specified in the constructor.
     // Deprecated: use the following overload instead.
-    virtual void OnMouseCursorPosition(CursorState state,
-                                       const DesktopVector& position) {}
+    virtual void OnMouseCursorPosition(CursorState /* state */,
+                                       const DesktopVector& /* position */) {}
 
     // Called in response to Capture(). `position` indicates cursor absolute
     // position on the system in fullscreen coordinate, i.e. the top-left
@@ -64,7 +64,7 @@ class MouseCursorMonitor {
     // TODO(zijiehe): Ensure all implementations return the absolute position.
     // TODO(zijiehe): Current this overload works correctly only when capturing
     // mouse cursor against fullscreen.
-    virtual void OnMouseCursorPosition(const DesktopVector& position) {}
+    virtual void OnMouseCursorPosition(const DesktopVector& /* position */) {}
 
    protected:
     virtual ~Callback() {}
diff --git a/modules/desktop_capture/window_capturer_unittest.cc b/modules/desktop_capture/window_capturer_unittest.cc
index 519c04601b..3483d4d2c8 100644
--- a/modules/desktop_capture/window_capturer_unittest.cc
+++ b/modules/desktop_capture/window_capturer_unittest.cc
@@ -33,7 +33,7 @@ class WindowCapturerTest : public ::testing::Test,
   void TearDown() override {}
 
   // DesktopCapturer::Callback interface
-  void OnCaptureResult(DesktopCapturer::Result result,
+  void OnCaptureResult(DesktopCapturer::Result /* result */,
                        std::unique_ptr<DesktopFrame> frame) override {
     frame_ = std::move(frame);
   }
diff --git a/modules/pacing/bitrate_prober.cc b/modules/pacing/bitrate_prober.cc
index dc9f4beab7..ba34546b2b 100644
--- a/modules/pacing/bitrate_prober.cc
+++ b/modules/pacing/bitrate_prober.cc
@@ -136,7 +136,7 @@ void BitrateProber::CreateProbeCluster(
                    << ")";
 }
 
-Timestamp BitrateProber::NextProbeTime(Timestamp now) const {
+Timestamp BitrateProber::NextProbeTime(Timestamp /* now */) const {
   // Probing is not active or probing is already complete.
   if (probing_state_ != ProbingState::kActive || clusters_.empty()) {
     return Timestamp::PlusInfinity();
diff --git a/modules/pacing/pacing_controller.h b/modules/pacing/pacing_controller.h
index 880af2917d..c9efed6b5a 100644
--- a/modules/pacing/pacing_controller.h
+++ b/modules/pacing/pacing_controller.h
@@ -57,9 +57,10 @@ class PacingController {
     // TODO(bugs.webrtc.org/11340): Make pure virtual once downstream projects
     // have been updated.
     virtual void OnAbortedRetransmissions(
-        uint32_t ssrc,
-        rtc::ArrayView<const uint16_t> sequence_numbers) {}
-    virtual std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
+        uint32_t /* ssrc */,
+        rtc::ArrayView<const uint16_t> /* sequence_numbers */) {}
+    virtual std::optional<uint32_t> GetRtxSsrcForMedia(
+        uint32_t /* ssrc */) const {
       return std::nullopt;
     }
   };
diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc
index 4618bd3c26..38af5c7b7d 100644
--- a/modules/pacing/pacing_controller_unittest.cc
+++ b/modules/pacing/pacing_controller_unittest.cc
@@ -106,7 +106,7 @@ class MediaStream {
 class MockPacingControllerCallback : public PacingController::PacketSender {
  public:
   void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
-                  const PacedPacketInfo& cluster_info) override {
+                  const PacedPacketInfo& /* cluster_info */) override {
     SendPacket(packet->Ssrc(), packet->SequenceNumber(),
                packet->capture_time().ms(),
                packet->packet_type() == RtpPacketMediaType::kRetransmission,
@@ -184,7 +184,7 @@ class PacingControllerPadding : public PacingController::PacketSender {
   PacingControllerPadding() : padding_sent_(0), total_bytes_sent_(0) {}
 
   void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
-                  const PacedPacketInfo& pacing_info) override {
+                  const PacedPacketInfo& /* pacing_info */) override {
     total_bytes_sent_ += packet->payload_size();
   }
 
@@ -1565,7 +1565,7 @@ TEST_F(PacingControllerTest, ProbeClusterId) {
   });
   bool non_probe_packet_seen = false;
   EXPECT_CALL(callback, SendPacket)
-      .WillOnce([&](std::unique_ptr<RtpPacketToSend> packet,
+      .WillOnce([&](std::unique_ptr<RtpPacketToSend> /* packet */,
                     const PacedPacketInfo& cluster_info) {
         EXPECT_EQ(cluster_info.probe_cluster_id, kNotAProbe);
         non_probe_packet_seen = true;
@@ -1647,7 +1647,7 @@ TEST_F(PacingControllerTest, SmallFirstProbePacket) {
 
   // Expect small padding packet to be requested.
   EXPECT_CALL(callback, GeneratePadding(DataSize::Bytes(1)))
-      .WillOnce([&](DataSize padding_size) {
+      .WillOnce([&](DataSize /* padding_size */) {
         std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
         padding_packets.emplace_back(
             BuildPacket(RtpPacketMediaType::kPadding, kAudioSsrc, 1,
@@ -1660,7 +1660,7 @@ TEST_F(PacingControllerTest, SmallFirstProbePacket) {
   EXPECT_CALL(callback, SendPacket)
       .Times(AnyNumber())
       .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
-                          const PacedPacketInfo& cluster_info) {
+                          const PacedPacketInfo& /* cluster_info */) {
         if (packets_sent == 0) {
           EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
         } else {
@@ -2205,7 +2205,7 @@ TEST_F(PacingControllerTest,
   size_t sent_size_in_burst = 0;
   EXPECT_CALL(callback, SendPacket)
       .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
-                          const PacedPacketInfo& cluster_info) {
+                          const PacedPacketInfo& /* cluster_info */) {
         sent_size_in_burst += packet->size();
       });
 
diff --git a/modules/pacing/packet_router_unittest.cc b/modules/pacing/packet_router_unittest.cc
index f9a55c7648..0d86f9e8a2 100644
--- a/modules/pacing/packet_router_unittest.cc
+++ b/modules/pacing/packet_router_unittest.cc
@@ -118,7 +118,7 @@ TEST_F(PacketRouterTest, GeneratePaddingPrioritizesRtx) {
   const size_t kExpectedPaddingPackets = 1;
   EXPECT_CALL(rtp_1, GeneratePadding(_)).Times(0);
   EXPECT_CALL(rtp_2, GeneratePadding(kPaddingSize))
-      .WillOnce([&](size_t padding_size) {
+      .WillOnce([&](size_t /* padding_size */) {
         return std::vector<std::unique_ptr<RtpPacketToSend>>(
             kExpectedPaddingPackets);
       });
@@ -163,7 +163,7 @@ TEST_F(PacketRouterTest, GeneratePaddingPrioritizesVideo) {
   const size_t kPaddingSize = 123;
   const size_t kExpectedPaddingPackets = 1;
 
-  auto generate_padding = [&](size_t padding_size) {
+  auto generate_padding = [&](size_t /* padding_size */) {
     return std::vector<std::unique_ptr<RtpPacketToSend>>(
         kExpectedPaddingPackets);
   };
@@ -267,7 +267,7 @@ TEST_F(PacketRouterTest, PadsOnLastActiveMediaStream) {
   // and supports rtx.
   EXPECT_CALL(rtp_2, GeneratePadding(kPaddingBytes))
       .Times(1)
-      .WillOnce([&](size_t target_size_bytes) {
+      .WillOnce([&](size_t /* target_size_bytes */) {
         std::vector<std::unique_ptr<RtpPacketToSend>> packets;
         packets.push_back(BuildRtpPacket(kSsrc2));
         return packets;
@@ -279,7 +279,7 @@ TEST_F(PacketRouterTest, PadsOnLastActiveMediaStream) {
 
   EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
       .Times(1)
-      .WillOnce([&](size_t target_size_bytes) {
+      .WillOnce([&](size_t /* target_size_bytes */) {
         std::vector<std::unique_ptr<RtpPacketToSend>> packets;
         packets.push_back(BuildRtpPacket(kSsrc1));
         return packets;