diff --git a/api/peer_connection_interface.cc b/api/peer_connection_interface.cc
index 6cdbf7023d..99e75d9b6e 100644
--- a/api/peer_connection_interface.cc
+++ b/api/peer_connection_interface.cc
@@ -104,13 +104,13 @@ PeerConnectionFactoryInterface::CreatePeerConnection(
 
 RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
 PeerConnectionFactoryInterface::CreatePeerConnectionOrError(
-    const PeerConnectionInterface::RTCConfiguration& configuration,
-    PeerConnectionDependencies dependencies) {
+    const PeerConnectionInterface::RTCConfiguration& /* configuration */,
+    PeerConnectionDependencies /* dependencies */) {
   return RTCError(RTCErrorType::INTERNAL_ERROR);
 }
 
 RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities(
-    cricket::MediaType kind) const {
+    cricket::MediaType /* kind */) const {
   return {};
 }
 
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index b9b6682e43..0108a69347 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -1352,7 +1352,7 @@ class PeerConnectionObserver {
   // RTCSessionDescription" algorithm:
   // https://w3c.github.io/webrtc-pc/#set-description
   virtual void OnTrack(
-      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
+      rtc::scoped_refptr<RtpTransceiverInterface> /* transceiver */) {}
 
   // Called when signaling indicates that media will no longer be received on a
   // track.
@@ -1363,7 +1363,7 @@ class PeerConnectionObserver {
   // https://w3c.github.io/webrtc-pc/#process-remote-track-removal
   // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
   virtual void OnRemoveTrack(
-      rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
+      rtc::scoped_refptr<RtpReceiverInterface> /* receiver */) {}
 
   // Called when an interesting usage is detected by WebRTC.
   // An appropriate action is to add information about the context of the
@@ -1371,7 +1371,7 @@ class PeerConnectionObserver {
   // log function.
   // The heuristics for defining what constitutes "interesting" are
   // implementation-defined.
-  virtual void OnInterestingUsage(int usage_pattern) {}
+  virtual void OnInterestingUsage(int /* usage_pattern */) {}
 };
 
 // PeerConnectionDependencies holds all of PeerConnections dependencies.
@@ -1615,7 +1615,7 @@ class RTC_EXPORT PeerConnectionFactoryInterface
   // StopAecDump function is called.
   // TODO(webrtc:6463): Delete default implementation when downstream mocks
   // classes are updated.
-  virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
+  virtual bool StartAecDump(FILE* /* file */, int64_t /* max_size_bytes */) {
     return false;
   }
 
diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc
index 2e1ac71e3d..b415b5fc34 100644
--- a/audio/channel_send_unittest.cc
+++ b/audio/channel_send_unittest.cc
@@ -298,7 +298,7 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToInsertedTransformedFrame) {
 
   std::optional<uint8_t> sent_audio_level;
   auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
-                      const PacketOptions& options) {
+                      const PacketOptions& /* options */) {
     RtpPacketReceived packet(&extension_manager);
     packet.Parse(data);
     RTPHeader header;
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 506abf42ef..7955331231 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -227,8 +227,9 @@ ConvertIceCandidatePairEventType(IceCandidatePairEventType type) {
 
 }  // namespace
 
-std::string RtcEventLogEncoderLegacy::EncodeLogStart(int64_t timestamp_us,
-                                                     int64_t utc_time_us) {
+std::string RtcEventLogEncoderLegacy::EncodeLogStart(
+    int64_t timestamp_us,
+    int64_t /* utc_time_us */) {
   rtclog::Event rtclog_event;
   rtclog_event.set_timestamp_us(timestamp_us);
   rtclog_event.set_type(rtclog::Event::LOG_START);
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/logging/rtc_event_log/rtc_event_log_unittest_helper.h
index 3d1fc115cb..0bcffdb489 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.h
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.h
@@ -248,8 +248,8 @@ class EventVerifier {
       const LoggedGenericAckReceived& logged_event) const;
 
   template <typename EventType, typename ParsedType>
-  void VerifyLoggedRtpPacket(const EventType& original_event,
-                             const ParsedType& logged_event) {
+  void VerifyLoggedRtpPacket(const EventType& /* original_event */,
+                             const ParsedType& /* logged_event */) {
     static_assert(sizeof(ParsedType) == 0,
                   "You have to use one of the two defined template "
                   "specializations of VerifyLoggedRtpPacket");
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 2584286ded..330a370f64 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -97,9 +97,10 @@ class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
   std::optional<uint32_t> GetUnsignaledSsrc() const override {
     return std::nullopt;
   }
-  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {}
+  void ChooseReceiverReportSsrc(
+      const std::set<uint32_t>& /* choices */) override {}
 
-  virtual bool SetLocalSsrc(const StreamParams& sp) { return true; }
+  virtual bool SetLocalSsrc(const StreamParams& /* sp */) { return true; }
   void OnDemuxerCriteriaUpdatePending() override {}
   void OnDemuxerCriteriaUpdateComplete() override {}
 
@@ -151,18 +152,19 @@ class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
   }
 
   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
-                            int64_t packet_time_us) {
+                            int64_t /* packet_time_us */) {
     rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
   }
 
-  void SetFrameDecryptor(uint32_t ssrc,
+  void SetFrameDecryptor(uint32_t /* ssrc */,
                          rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
-                             frame_decryptor) override {}
+                         /* frame_decryptor */) override {}
 
   void SetDepacketizerToDecoderFrameTransformer(
-      uint32_t ssrc,
-      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
-      override {}
+      uint32_t /* ssrc */,
+      rtc::scoped_refptr<
+          webrtc::FrameTransformerInterface> /* frame_transformer */) override {
+  }
 
   void SetInterface(MediaChannelNetworkInterface* iface) override {
     network_interface_ = iface;
@@ -363,18 +365,19 @@ class RtpSendChannelHelper : public Base, public MediaChannelUtil {
   }
 
   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
-                            int64_t packet_time_us) {
+                            int64_t /* packet_time_us */) {
     rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
   }
 
   // Stuff that deals with encryptors, transformers and the like
-  void SetFrameEncryptor(uint32_t ssrc,
+  void SetFrameEncryptor(uint32_t /* ssrc */,
                          rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
-                             frame_encryptor) override {}
+                         /* frame_encryptor */) override {}
   void SetEncoderToPacketizerFrameTransformer(
-      uint32_t ssrc,
-      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
-      override {}
+      uint32_t /* ssrc */,
+      rtc::scoped_refptr<
+          webrtc::FrameTransformerInterface> /* frame_transformer */) override {
+  }
 
   void SetInterface(MediaChannelNetworkInterface* iface) override {
     network_interface_ = iface;
@@ -407,9 +410,9 @@ class RtpSendChannelHelper : public Base, public MediaChannelUtil {
   void set_send_rtcp_parameters(const RtcpParameters& params) {
     send_rtcp_parameters_ = params;
   }
-  void OnPacketSent(const rtc::SentPacket& sent_packet) override {}
+  void OnPacketSent(const rtc::SentPacket& /* sent_packet */) override {}
   void OnReadyToSend(bool ready) override { ready_to_send_ = ready; }
-  void OnNetworkRouteChanged(absl::string_view transport_name,
+  void OnNetworkRouteChanged(absl::string_view /* transport_name */,
                              const rtc::NetworkRoute& network_route) override {
     last_network_route_ = network_route;
     ++num_network_route_changes_;
@@ -496,9 +499,9 @@ class FakeVoiceMediaReceiveChannel
       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
-  void SetReceiveNackEnabled(bool enabled) override {}
-  void SetRtcpMode(webrtc::RtcpMode mode) override {}
-  void SetReceiveNonSenderRttEnabled(bool enabled) override {}
+  void SetReceiveNackEnabled(bool /* enabled */) override {}
+  void SetRtcpMode(webrtc::RtcpMode /* mode */) override {}
+  void SetReceiveNonSenderRttEnabled(bool /* enabled */) override {}
 
  private:
   class VoiceChannelAudioSink : public AudioSource::Sink {
@@ -574,8 +577,8 @@ class FakeVoiceMediaSendChannel
 
   bool SenderNackEnabled() const override { return false; }
   bool SenderNonSenderRttEnabled() const override { return false; }
-  void SetReceiveNackEnabled(bool enabled) {}
-  void SetReceiveNonSenderRttEnabled(bool enabled) {}
+  void SetReceiveNackEnabled(bool /* enabled */) {}
+  void SetReceiveNonSenderRttEnabled(bool /* enabled */) {}
   bool SendCodecHasNack() const override { return false; }
   void SetSendCodecChangedCallback(
       absl::AnyInvocable<void()> callback) override {}
diff --git a/media/base/fake_network_interface.h b/media/base/fake_network_interface.h
index d0763fe533..c99282a6c9 100644
--- a/media/base/fake_network_interface.h
+++ b/media/base/fake_network_interface.h
@@ -154,7 +154,9 @@ class FakeNetworkInterface : public MediaChannelNetworkInterface {
     return true;
   }
 
-  virtual int SetOption(SocketType type, rtc::Socket::Option opt, int option) {
+  virtual int SetOption(SocketType /* type */,
+                        rtc::Socket::Option opt,
+                        int option) {
     if (opt == rtc::Socket::OPT_SNDBUF) {
       sendbuf_size_ = option;
     } else if (opt == rtc::Socket::OPT_RCVBUF) {
diff --git a/media/engine/fake_video_codec_factory.cc b/media/engine/fake_video_codec_factory.cc
index 9b544fb2b9..1b62cff10c 100644
--- a/media/engine/fake_video_codec_factory.cc
+++ b/media/engine/fake_video_codec_factory.cc
@@ -39,7 +39,7 @@ std::vector<SdpVideoFormat> FakeVideoEncoderFactory::GetSupportedFormats()
 
 std::unique_ptr<VideoEncoder> FakeVideoEncoderFactory::Create(
     const Environment& env,
-    const SdpVideoFormat& format) {
+    const SdpVideoFormat& /* format */) {
   return std::make_unique<test::FakeEncoder>(env);
 }
 
@@ -57,8 +57,8 @@ std::vector<SdpVideoFormat> FakeVideoDecoderFactory::GetSupportedFormats()
 }
 
 std::unique_ptr<VideoDecoder> FakeVideoDecoderFactory::Create(
-    const Environment& env,
-    const SdpVideoFormat& format) {
+    const Environment& /* env */,
+    const SdpVideoFormat& /* format */) {
   return std::make_unique<test::FakeDecoder>();
 }
 
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index c4e5f615a2..38deaff1f6 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -97,8 +97,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
                    webrtc::SetParametersCallback callback) override;
   void Start() override { sending_ = true; }
   void Stop() override { sending_ = false; }
-  void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
-  }
+  void SendAudioData(
+      std::unique_ptr<webrtc::AudioFrame> /* audio_frame */) override {}
   bool SendTelephoneEvent(int payload_type,
                           int payload_frequency,
                           int event,
@@ -301,14 +301,16 @@ class FakeVideoReceiveStream final
   void UpdateRtxSsrc(uint32_t ssrc) { config_.rtp.rtx_ssrc = ssrc; }
 
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
-                             frame_decryptor) override {}
+                         /* frame_decryptor */) override {}
 
   void SetDepacketizerToDecoderFrameTransformer(
-      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
-      override {}
+      rtc::scoped_refptr<
+          webrtc::FrameTransformerInterface> /* frame_transformer */) override {
+  }
 
-  RecordingState SetAndGetRecordingState(RecordingState state,
-                                         bool generate_key_frame) override {
+  RecordingState SetAndGetRecordingState(
+      RecordingState /* state */,
+      bool /* generate_key_frame */) override {
     return RecordingState();
   }
   void GenerateKeyFrame() override {}
@@ -400,14 +402,14 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
 class FakePayloadTypeSuggester : public webrtc::PayloadTypeSuggester {
  public:
   webrtc::RTCErrorOr<webrtc::PayloadType> SuggestPayloadType(
-      const std::string& mid,
+      const std::string& /* mid */,
       cricket::Codec codec) override {
     // Ignores mid argument.
     return pt_picker_.SuggestMapping(codec, nullptr);
   }
-  webrtc::RTCError AddLocalMapping(const std::string& mid,
-                                   webrtc::PayloadType payload_type,
-                                   const cricket::Codec& codec) override {
+  webrtc::RTCError AddLocalMapping(const std::string& /* mid */,
+                                   webrtc::PayloadType /* payload_type */,
+                                   const cricket::Codec& /* codec */) override {
     return webrtc::RTCError::OK();
   }
 
diff --git a/media/engine/fake_webrtc_video_engine.cc b/media/engine/fake_webrtc_video_engine.cc
index a701a9565c..d13155ffdf 100644
--- a/media/engine/fake_webrtc_video_engine.cc
+++ b/media/engine/fake_webrtc_video_engine.cc
@@ -58,7 +58,7 @@ FakeWebRtcVideoDecoder::~FakeWebRtcVideoDecoder() {
   }
 }
 
-bool FakeWebRtcVideoDecoder::Configure(const Settings& settings) {
+bool FakeWebRtcVideoDecoder::Configure(const Settings& /* settings */) {
   return true;
 }
 
@@ -99,7 +99,7 @@ FakeWebRtcVideoDecoderFactory::GetSupportedFormats() const {
 }
 
 std::unique_ptr<webrtc::VideoDecoder> FakeWebRtcVideoDecoderFactory::Create(
-    const webrtc::Environment& env,
+    const webrtc::Environment& /* env */,
     const webrtc::SdpVideoFormat& format) {
   if (format.IsCodecInList(supported_codec_formats_)) {
     num_created_decoders_++;
@@ -147,13 +147,13 @@ FakeWebRtcVideoEncoder::~FakeWebRtcVideoEncoder() {
 }
 
 void FakeWebRtcVideoEncoder::SetFecControllerOverride(
-    webrtc::FecControllerOverride* fec_controller_override) {
+    webrtc::FecControllerOverride* /* fec_controller_override */) {
   // Ignored.
 }
 
 int32_t FakeWebRtcVideoEncoder::InitEncode(
     const webrtc::VideoCodec* codecSettings,
-    const VideoEncoder::Settings& settings) {
+    const VideoEncoder::Settings& /* settings */) {
   webrtc::MutexLock lock(&mutex_);
   codec_settings_ = *codecSettings;
   init_encode_event_.Set();
@@ -161,8 +161,8 @@ int32_t FakeWebRtcVideoEncoder::InitEncode(
 }
 
 int32_t FakeWebRtcVideoEncoder::Encode(
-    const webrtc::VideoFrame& inputImage,
-    const std::vector<webrtc::VideoFrameType>* frame_types) {
+    const webrtc::VideoFrame& /* inputImage */,
+    const std::vector<webrtc::VideoFrameType>* /* frame_types */) {
   webrtc::MutexLock lock(&mutex_);
   ++num_frames_encoded_;
   init_encode_event_.Set();
@@ -170,7 +170,7 @@ int32_t FakeWebRtcVideoEncoder::Encode(
 }
 
 int32_t FakeWebRtcVideoEncoder::RegisterEncodeCompleteCallback(
-    webrtc::EncodedImageCallback* callback) {
+    webrtc::EncodedImageCallback* /* callback */) {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
@@ -178,8 +178,8 @@ int32_t FakeWebRtcVideoEncoder::Release() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-void FakeWebRtcVideoEncoder::SetRates(const RateControlParameters& parameters) {
-}
+void FakeWebRtcVideoEncoder::SetRates(
+    const RateControlParameters& /* parameters */) {}
 
 webrtc::VideoEncoder::EncoderInfo FakeWebRtcVideoEncoder::GetEncoderInfo()
     const {
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index 7d3329f8de..7195a39957 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -707,7 +707,7 @@ EncodedImageCallback::Result SimulcastEncoderAdapter::OnEncodedImage(
                                                     &stream_codec_specific);
 }
 
-void SimulcastEncoderAdapter::OnDroppedFrame(size_t stream_idx) {
+void SimulcastEncoderAdapter::OnDroppedFrame(size_t /* stream_idx */) {
   // Not yet implemented.
 }
 
diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc
index 8baccfd4f5..b5c4cd1a5c 100644
--- a/media/engine/simulcast_encoder_adapter_unittest.cc
+++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -58,13 +58,13 @@ std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture(
   std::unique_ptr<VideoEncoderFactory> encoder_factory =
       std::make_unique<FunctionVideoEncoderFactory>(
           [internal_encoder_factory](const Environment& env,
-                                     const SdpVideoFormat& format) {
+                                     const SdpVideoFormat& /* format */) {
             return std::make_unique<SimulcastEncoderAdapter>(
                 env, internal_encoder_factory, nullptr, SdpVideoFormat::VP8());
           });
   std::unique_ptr<VideoDecoderFactory> decoder_factory =
       std::make_unique<FunctionVideoDecoderFactory>(
-          [](const Environment& env, const SdpVideoFormat& format) {
+          [](const Environment& env, const SdpVideoFormat& /* format */) {
             return CreateVp8Decoder(env);
           });
   return CreateSimulcastTestFixture(std::move(encoder_factory),
@@ -223,7 +223,7 @@ class MockVideoEncoder : public VideoEncoder {
               (override));
 
   int32_t InitEncode(const VideoCodec* codecSettings,
-                     const VideoEncoder::Settings& settings) override {
+                     const VideoEncoder::Settings& /* settings */) override {
     codec_ = *codecSettings;
     if (codec_.numberOfSimulcastStreams > 1 && fallback_from_simulcast_) {
       return *fallback_from_simulcast_;
@@ -374,7 +374,7 @@ std::vector<SdpVideoFormat> MockVideoEncoderFactory::GetSupportedFormats()
 }
 
 std::unique_ptr<VideoEncoder> MockVideoEncoderFactory::Create(
-    const Environment& env,
+    const Environment& /* env */,
     const SdpVideoFormat& format) {
   if (create_video_encoder_return_nullptr_) {
     return nullptr;
@@ -480,8 +480,9 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
     SetUp();
   }
 
-  Result OnEncodedImage(const EncodedImage& encoded_image,
-                        const CodecSpecificInfo* codec_specific_info) override {
+  Result OnEncodedImage(
+      const EncodedImage& encoded_image,
+      const CodecSpecificInfo* /* codec_specific_info */) override {
     last_encoded_image_width_ = encoded_image._encodedWidth;
     last_encoded_image_height_ = encoded_image._encodedHeight;
     last_encoded_image_simulcast_index_ = encoded_image.SimulcastIndex();
@@ -1146,7 +1147,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, NativeHandleForwardingOnlyIfSupported) {
   // ...the lowest one gets a software buffer.
   EXPECT_CALL(*encoders[0], Encode)
       .WillOnce([&](const VideoFrame& frame,
-                    const std::vector<VideoFrameType>* frame_types) {
+                    const std::vector<VideoFrameType>* /* frame_types */) {
         EXPECT_EQ(frame.video_frame_buffer()->type(),
                   VideoFrameBuffer::Type::kI420);
         return 0;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 0f83d92bd3..e3b99f23a3 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -288,7 +288,7 @@ webrtc::AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
     bool use_nack,
     bool enable_non_sender_rtt,
     const std::vector<std::string>& stream_ids,
-    const std::vector<webrtc::RtpExtension>& extensions,
+    const std::vector<webrtc::RtpExtension>& /* extensions */,
     webrtc::Transport* rtcp_send_transport,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc
index ae8d778189..9d35fc236a 100644
--- a/media/sctp/dcsctp_transport.cc
+++ b/media/sctp/dcsctp_transport.cc
@@ -685,7 +685,7 @@ void DcSctpTransport::OnTransportWritableState(
 }
 
 void DcSctpTransport::OnTransportReadPacket(
-    rtc::PacketTransportInternal* transport,
+    rtc::PacketTransportInternal* /* transport */,
     const rtc::ReceivedPacket& packet) {
   RTC_DCHECK_RUN_ON(network_thread_);
   if (packet.decryption_info() != rtc::ReceivedPacket::kDtlsDecrypted) {
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index c08fd4d661..c022f01d38 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -119,12 +119,13 @@ std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
 }
 
 // This method receives the callback from ACM when a new packet is produced.
-int32_t AcmSendTestOldApi::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) {
+int32_t AcmSendTestOldApi::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 */) {
   // Store the packet locally.
   frame_type_ = frame_type;
   payload_type_ = payload_type;
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 56726cfad2..e3523cd29d 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -114,7 +114,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
                    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 {
     MutexLock lock(&mutex_);
     ++num_calls_;
     last_frame_type_ = frame_type;
@@ -1054,14 +1054,14 @@ class AcmSetBitRateTest : public ::testing::Test {
                                  int channels,
                                  int payload_type,
                                  int frame_size_samples,
-                                 int frame_size_rtp_timestamps) {
+                                 int /* frame_size_rtp_timestamps */) {
     return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
                                      payload_type, frame_size_samples);
   }
 
   void RegisterExternalSendCodec(
       std::unique_ptr<AudioEncoder> external_speech_encoder,
-      int payload_type) {
+      int /* payload_type */) {
     send_test_->RegisterExternalCodec(std::move(external_speech_encoder));
   }
 
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index c47434f9aa..c3ec2d7072 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -97,7 +97,7 @@ bool FrameLengthController::Config::FrameLengthChange::operator<(
 }
 
 bool FrameLengthController::FrameLengthIncreasingDecision(
-    const AudioEncoderRuntimeConfig& config) {
+    const AudioEncoderRuntimeConfig& /* config */) {
   // Increase frame length if
   // 1. `uplink_bandwidth_bps` is known to be smaller or equal than
   //    `min_encoder_bitrate_bps` plus `prevent_overuse_margin_bps` plus the
@@ -153,7 +153,7 @@ bool FrameLengthController::FrameLengthIncreasingDecision(
 }
 
 bool FrameLengthController::FrameLengthDecreasingDecision(
-    const AudioEncoderRuntimeConfig& config) {
+    const AudioEncoderRuntimeConfig& /* config */) {
   // Decrease frame length if
   // 1. shorter frame length is available AND
   // 2. `uplink_bandwidth_bps` is known to be bigger than
diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index ff7e919d9b..c80c572e27 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -52,7 +52,7 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
-int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+int AudioDecoderPcmU::PacketDuration(const uint8_t* /* encoded */,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
@@ -98,7 +98,7 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
-int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+int AudioDecoderPcmA::PacketDuration(const uint8_t* /* encoded */,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index bca47cea13..85b3b58e30 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -57,7 +57,7 @@ std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
                                                  timestamp, 8, 16);
 }
 
-int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
+int AudioDecoderG722Impl::PacketDuration(const uint8_t* /* encoded */,
                                          size_t encoded_len) const {
   // 1/2 encoded byte per sample per channel.
   return static_cast<int>(2 * encoded_len / Channels());
@@ -125,7 +125,7 @@ int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
-int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* encoded,
+int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* /* encoded */,
                                                size_t encoded_len) const {
   // 1/2 encoded byte per sample per channel. Make sure the length represents
   // an equal number of bytes per channel. Otherwise, we cannot de-interleave
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 9eed1c645d..6d12773c57 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -130,7 +130,7 @@ size_t AudioDecoderOpusImpl::Channels() const {
 }
 
 void AudioDecoderOpusImpl::GeneratePlc(
-    size_t requested_samples_per_channel,
+    size_t /* requested_samples_per_channel */,
     rtc::BufferT<int16_t>* concealment_audio) {
   if (!generate_plc_) {
     return;
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index fc37a219a4..12ff7b5193 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -65,7 +65,7 @@ std::unique_ptr<AudioEncoderOpusStates> CreateCodec(
 
   MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
   AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
-      [mock_ptr](absl::string_view, RtcEventLog* event_log) {
+      [mock_ptr](absl::string_view, RtcEventLog* /* event_log */) {
         std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
             new NiceMock<MockAudioNetworkAdaptor>());
         EXPECT_CALL(*adaptor, Die());
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 2e05447ed6..b332326061 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -48,8 +48,8 @@ class PowerRatioEstimator : public LappedTransform::Callback {
  protected:
   void ProcessAudioBlock(const std::complex<float>* const* input,
                          size_t num_input_channels,
-                         size_t num_freq_bins,
-                         size_t num_output_channels,
+                         size_t /* num_freq_bins */,
+                         size_t /* num_output_channels */,
                          std::complex<float>* const* output) override {
     float low_pow = 0.f;
     float high_pow = 0.f;