Comment unused variables in implemented functions
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
Bug: webrtc:370878648
Change-Id: I0ab2eafd26e46312e4595f302b92006c9e23d5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364340
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43157}
diff --git a/api/array_view.h b/api/array_view.h
index 4abd2bf..f1eb6c1 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -97,7 +97,7 @@
static_assert(Size > 0, "ArrayView size must be variable or non-negative");
public:
- ArrayViewBase(T* data, size_t size) : data_(data) {}
+ ArrayViewBase(T* data, size_t /* size */) : data_(data) {}
static constexpr size_t size() { return Size; }
static constexpr bool empty() { return false; }
@@ -114,7 +114,7 @@
template <typename T>
class ArrayViewBase<T, 0> {
public:
- explicit ArrayViewBase(T* data, size_t size) {}
+ explicit ArrayViewBase(T* /* data */, size_t /* size */) {}
static constexpr size_t size() { return 0; }
static constexpr bool empty() { return true; }
diff --git a/api/dtmf_sender_interface.h b/api/dtmf_sender_interface.h
index 4692f25..16ce665 100644
--- a/api/dtmf_sender_interface.h
+++ b/api/dtmf_sender_interface.h
@@ -27,12 +27,12 @@
// tones.
// The callback includes the state of the tone buffer at the time when
// the tone finished playing.
- virtual void OnToneChange(const std::string& tone,
- const std::string& tone_buffer) {}
+ virtual void OnToneChange(const std::string& /* tone */,
+ const std::string& /* tone_buffer */) {}
// DEPRECATED: Older API without tone buffer.
// TODO(bugs.webrtc.org/9725): Remove old API and default implementation
// when old callers are gone.
- virtual void OnToneChange(const std::string& tone) {}
+ virtual void OnToneChange(const std::string& /* tone */) {}
protected:
virtual ~DtmfSenderObserverInterface() = default;
@@ -91,7 +91,7 @@
virtual bool InsertDtmf(const std::string& tones,
int duration,
int inter_tone_gap,
- int comma_delay) {
+ int /* comma_delay */) {
// TODO(bugs.webrtc.org/165700): Remove once downstream implementations
// override this signature rather than the 3-parameter one.
return InsertDtmf(tones, duration, inter_tone_gap);
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index 6959445..3f41336 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -139,9 +139,9 @@
rtc::scoped_refptr<TransformedFrameCallback>) {}
virtual void RegisterTransformedFrameSinkCallback(
rtc::scoped_refptr<TransformedFrameCallback>,
- uint32_t ssrc) {}
+ uint32_t /* ssrc */) {}
virtual void UnregisterTransformedFrameCallback() {}
- virtual void UnregisterTransformedFrameSinkCallback(uint32_t ssrc) {}
+ virtual void UnregisterTransformedFrameSinkCallback(uint32_t /* ssrc */) {}
protected:
~FrameTransformerInterface() override = default;
diff --git a/api/function_view.h b/api/function_view.h
index b191a05..f853ddb 100644
--- a/api/function_view.h
+++ b/api/function_view.h
@@ -86,7 +86,7 @@
typename std::enable_if<std::is_same<
std::nullptr_t,
typename std::remove_cv<F>::type>::value>::type* = nullptr>
- FunctionView(F&& f) : call_(nullptr) {}
+ FunctionView(F&& /* f */) : call_(nullptr) {}
// Default constructor. Creates an empty FunctionView.
FunctionView() : call_(nullptr) {}
diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index d7c0eda..5a1e298 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -161,7 +161,7 @@
// The call is expected to happen on the network thread.
// TODO(crbug/1255737): make pure virtual once downstream project adapts.
virtual void ProcessConstraints(
- const webrtc::VideoTrackSourceConstraints& constraints) {}
+ const webrtc::VideoTrackSourceConstraints& /* constraints */) {}
protected:
~VideoTrackSourceInterface() override = default;
@@ -184,14 +184,14 @@
// Register a video sink for this track. Used to connect the track to the
// underlying video engine.
- void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
- const rtc::VideoSinkWants& wants) override {}
- void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
+ void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* /* sink */,
+ const rtc::VideoSinkWants& /* wants */) override {}
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* /* sink */) override {}
virtual VideoTrackSourceInterface* GetSource() const = 0;
virtual ContentHint content_hint() const;
- virtual void set_content_hint(ContentHint hint) {}
+ virtual void set_content_hint(ContentHint /* hint */) {}
protected:
~VideoTrackInterface() override = default;
@@ -200,11 +200,11 @@
// Interface for receiving audio data from a AudioTrack.
class AudioTrackSinkInterface {
public:
- virtual void OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) {
+ virtual void OnData(const void* /* audio_data */,
+ int /* bits_per_sample */,
+ int /* sample_rate */,
+ size_t /* number_of_channels */,
+ size_t /* number_of_frames */) {
RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
}
@@ -212,12 +212,13 @@
// supposed to deliver the timestamp when this audio frame was originally
// captured. This timestamp MUST be based on the same clock as
// rtc::TimeMillis().
- virtual void 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) {
+ virtual void 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 */) {
// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
// pure virtual.
return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
@@ -251,15 +252,15 @@
// Sets the volume of the source. `volume` is in the range of [0, 10].
// TODO(tommi): This method should be on the track and ideally volume should
// be applied in the track in a way that does not affect clones of the track.
- virtual void SetVolume(double volume) {}
+ virtual void SetVolume(double /* volume */) {}
// Registers/unregisters observers to the audio source.
- virtual void RegisterAudioObserver(AudioObserver* observer) {}
- virtual void UnregisterAudioObserver(AudioObserver* observer) {}
+ virtual void RegisterAudioObserver(AudioObserver* /* observer */) {}
+ virtual void UnregisterAudioObserver(AudioObserver* /* observer */) {}
// TODO(tommi): Make pure virtual.
- virtual void AddSink(AudioTrackSinkInterface* sink) {}
- virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
+ virtual void AddSink(AudioTrackSinkInterface* /* sink */) {}
+ virtual void RemoveSink(AudioTrackSinkInterface* /* sink */) {}
// Returns options for the AudioSource.
// (for some of the settings this approach is broken, e.g. setting
@@ -339,16 +340,18 @@
// Note: Default implementations are for avoiding link time errors in
// implementations that mock this API.
// TODO(bugs.webrtc.org/13980): Remove default implementations.
- virtual bool AddTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
+ virtual bool AddTrack(rtc::scoped_refptr<AudioTrackInterface> /* track */) {
RTC_CHECK_NOTREACHED();
}
- virtual bool AddTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
+ virtual bool AddTrack(rtc::scoped_refptr<VideoTrackInterface> /* track */) {
RTC_CHECK_NOTREACHED();
}
- virtual bool RemoveTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
+ virtual bool RemoveTrack(
+ rtc::scoped_refptr<AudioTrackInterface> /* track */) {
RTC_CHECK_NOTREACHED();
}
- virtual bool RemoveTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
+ virtual bool RemoveTrack(
+ rtc::scoped_refptr<VideoTrackInterface> /* track */) {
RTC_CHECK_NOTREACHED();
}
diff --git a/api/rtc_event_log/rtc_event_log.h b/api/rtc_event_log/rtc_event_log.h
index 3bd2553..55dee7d 100644
--- a/api/rtc_event_log/rtc_event_log.h
+++ b/api/rtc_event_log/rtc_event_log.h
@@ -60,7 +60,7 @@
bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) override;
void StopLogging() override {}
- void Log(std::unique_ptr<RtcEvent> event) override {}
+ void Log(std::unique_ptr<RtcEvent> /* event */) override {}
};
} // namespace webrtc
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index f97ed53..bad3d2c 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -80,7 +80,9 @@
virtual RtpParameters GetParameters() const = 0;
// TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
// Currently, doesn't support changing any parameters.
- virtual bool SetParameters(const RtpParameters& parameters) { return false; }
+ virtual bool SetParameters(const RtpParameters& /* parameters */) {
+ return false;
+ }
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index aa2eb6f..9afe3e3 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -119,7 +119,7 @@
// Default implementation of SetFrameTransformer.
// TODO: bugs.webrtc.org/15929 - remove when all implementations are good
void SetFrameTransformer(rtc::scoped_refptr<FrameTransformerInterface>
- frame_transformer) override {}
+ /* frame_transformer */) override {}
protected:
~RtpSenderInterface() override = default;
diff --git a/api/transport/data_channel_transport_interface.h b/api/transport/data_channel_transport_interface.h
index ddf6147..7ea1d30 100644
--- a/api/transport/data_channel_transport_interface.h
+++ b/api/transport/data_channel_transport_interface.h
@@ -88,7 +88,7 @@
// Callback issued when the data channel becomes unusable (closed).
// TODO(https://crbug.com/webrtc/10360): Make pure virtual when all
// consumers updated.
- virtual void OnTransportClosed(RTCError error) {}
+ virtual void OnTransportClosed(RTCError /* error */) {}
// The data channel's buffered_amount has fallen to or below the threshold
// set when calling `SetBufferedAmountLowThreshold`
diff --git a/api/video/video_sink_interface.h b/api/video/video_sink_interface.h
index f661248..5a1f67b 100644
--- a/api/video/video_sink_interface.h
+++ b/api/video/video_sink_interface.h
@@ -29,7 +29,7 @@
// Called on the network thread when video constraints change.
// TODO(crbug/1255737): make pure virtual once downstream project adapts.
virtual void OnConstraintsChanged(
- const webrtc::VideoTrackSourceConstraints& constraints) {}
+ const webrtc::VideoTrackSourceConstraints& /* constraints */) {}
};
} // namespace rtc
diff --git a/api/video_codecs/video_decoder.h b/api/video_codecs/video_decoder.h
index 934e224..079753d 100644
--- a/api/video_codecs/video_decoder.h
+++ b/api/video_codecs/video_decoder.h
@@ -107,7 +107,7 @@
// TODO(bugs.webrtc.org/15444): Migrate all subclasses to Decode() without
// missing_frame and delete this.
virtual int32_t Decode(const EncodedImage& input_image,
- bool missing_frames,
+ bool /* missing_frames */,
int64_t render_time_ms) {
return Decode(input_image, render_time_ms);
}
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index aadf452..44a7516 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -82,7 +82,7 @@
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) = 0;
- virtual void OnDroppedFrame(DropReason reason) {}
+ virtual void OnDroppedFrame(DropReason /* reason */) {}
};
class RTC_EXPORT VideoEncoder {
diff --git a/api/video_codecs/video_encoder_factory.h b/api/video_codecs/video_encoder_factory.h
index cb00f92..8986e92 100644
--- a/api/video_codecs/video_encoder_factory.h
+++ b/api/video_codecs/video_encoder_factory.h
@@ -54,7 +54,7 @@
// Called every time the encoder input resolution change. Should return a
// non-empty if an encoder switch should be performed.
virtual std::optional<SdpVideoFormat> OnResolutionChange(
- const RenderResolution& resolution) {
+ const RenderResolution& /* resolution */) {
return std::nullopt;
}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 73d2b25..483abdf 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -246,9 +246,9 @@
// note: The encoder_selector object must remain valid for the lifetime of the
// MediaChannel, unless replaced.
virtual void SetEncoderSelector(
- uint32_t ssrc,
- webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
- }
+ uint32_t /* ssrc */,
+ webrtc::VideoEncoderFactory::
+ EncoderSelectorInterface* /* encoder_selector */) {}
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
virtual bool SendCodecHasNack() const = 0;
// Called whenever the list of sending SSRCs changes.
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index eb4d1d0..c9d1f17 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -168,20 +168,22 @@
public:
virtual ~NetworkLinkRtcpObserver() = default;
- virtual void OnTransportFeedback(Timestamp receive_time,
- const rtcp::TransportFeedback& feedback) {}
+ virtual void OnTransportFeedback(
+ Timestamp /* receive_time */,
+ const rtcp::TransportFeedback& /* feedback */) {}
// RFC 8888 congestion control feedback.
virtual void OnCongestionControlFeedback(
- Timestamp receive_time,
- const rtcp::CongestionControlFeedback& feedback) {}
- virtual void OnReceiverEstimatedMaxBitrate(Timestamp receive_time,
- DataRate bitrate) {}
+ Timestamp /* receive_time */,
+ const rtcp::CongestionControlFeedback& /* feedback */) {}
+ virtual void OnReceiverEstimatedMaxBitrate(Timestamp /* receive_time */,
+ DataRate /* bitrate */) {}
// Called on an RTCP packet with sender or receiver reports with non zero
// report blocks. Report blocks are combined from all reports into one array.
- virtual void OnReport(Timestamp receive_time,
- rtc::ArrayView<const ReportBlockData> report_blocks) {}
- virtual void OnRttUpdate(Timestamp receive_time, TimeDelta rtt) {}
+ virtual void OnReport(
+ Timestamp /* receive_time */,
+ rtc::ArrayView<const ReportBlockData> /* report_blocks */) {}
+ virtual void OnRttUpdate(Timestamp /* receive_time */, TimeDelta /* rtt */) {}
};
// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
diff --git a/rtc_base/checks.cc b/rtc_base/checks.cc
index e732a26..14cfe69 100644
--- a/rtc_base/checks.cc
+++ b/rtc_base/checks.cc
@@ -78,8 +78,8 @@
abort();
}
-RTC_NORETURN void WriteFatalLog(const char* file,
- int line,
+RTC_NORETURN void WriteFatalLog(const char* /* file */,
+ int /* line */,
absl::string_view output) {
WriteFatalLog(output);
}
diff --git a/rtc_base/crypto_random.cc b/rtc_base/crypto_random.cc
index 68b86a0..d9660b1 100644
--- a/rtc_base/crypto_random.cc
+++ b/rtc_base/crypto_random.cc
@@ -34,7 +34,7 @@
public:
SecureRandomGenerator() {}
~SecureRandomGenerator() override {}
- bool Init(const void* seed, size_t len) override { return true; }
+ bool Init(const void* /* seed */, size_t /* len */) override { return true; }
bool Generate(void* buf, size_t len) override {
return (RAND_bytes(reinterpret_cast<unsigned char*>(buf), len) > 0);
}
@@ -45,7 +45,7 @@
public:
TestRandomGenerator() : seed_(7) {}
~TestRandomGenerator() override {}
- bool Init(const void* seed, size_t len) override { return true; }
+ bool Init(const void* /* seed */, size_t /* len */) override { return true; }
bool Generate(void* buf, size_t len) override {
for (size_t i = 0; i < len; ++i) {
static_cast<uint8_t*>(buf)[i] = static_cast<uint8_t>(GetRandom());
diff --git a/rtc_base/logging.cc b/rtc_base/logging.cc
index 61a3c66..f0461f7 100644
--- a/rtc_base/logging.cc
+++ b/rtc_base/logging.cc
@@ -220,7 +220,7 @@
}
}
-void LogMessage::AddTag(const char* tag) {
+void LogMessage::AddTag([[maybe_unused]] const char* tag) {
#ifdef WEBRTC_ANDROID
log_line_.set_tag(tag);
#endif
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 208e8b9..2b47ee8 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -466,7 +466,7 @@
// This has to be an operator with a precedence lower than << but
// higher than ?:
template <typename... Ts>
- void operator&(LogStreamer<Ts...>&& streamer) {}
+ void operator&(LogStreamer<Ts...>&& /* streamer */) {}
};
} // namespace webrtc_logging_impl
diff --git a/rtc_base/numerics/safe_conversions_impl.h b/rtc_base/numerics/safe_conversions_impl.h
index e924ce3..fe0f0d7 100644
--- a/rtc_base/numerics/safe_conversions_impl.h
+++ b/rtc_base/numerics/safe_conversions_impl.h
@@ -101,7 +101,9 @@
// Dst range always contains the result: nothing to check.
template <typename Dst, typename Src, DstSign IsDstSigned, SrcSign IsSrcSigned>
struct RangeCheckImpl<Dst, Src, IsDstSigned, IsSrcSigned, CONTAINS_RANGE> {
- static constexpr RangeCheckResult Check(Src value) { return TYPE_VALID; }
+ static constexpr RangeCheckResult Check(Src /* value */) {
+ return TYPE_VALID;
+ }
};
// Signed to signed narrowing.
diff --git a/rtc_base/sanitizer.h b/rtc_base/sanitizer.h
index 8af0824..08fc55e 100644
--- a/rtc_base/sanitizer.h
+++ b/rtc_base/sanitizer.h
@@ -51,9 +51,9 @@
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being unaddressable, so that reads and writes are not allowed. ASan may
// narrow the range to the nearest alignment boundaries.
-static inline void rtc_AsanPoison(const volatile void* ptr,
- size_t element_size,
- size_t num_elements) {
+static inline void rtc_AsanPoison([[maybe_unused]] const volatile void* ptr,
+ [[maybe_unused]] size_t element_size,
+ [[maybe_unused]] size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
@@ -62,9 +62,9 @@
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being addressable, so that reads and writes are allowed. ASan may widen
// the range to the nearest alignment boundaries.
-static inline void rtc_AsanUnpoison(const volatile void* ptr,
- size_t element_size,
- size_t num_elements) {
+static inline void rtc_AsanUnpoison([[maybe_unused]] const volatile void* ptr,
+ [[maybe_unused]] size_t element_size,
+ [[maybe_unused]] size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
@@ -72,9 +72,10 @@
// Ask MSan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being uninitialized.
-static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
- size_t element_size,
- size_t num_elements) {
+static inline void rtc_MsanMarkUninitialized(
+ [[maybe_unused]] const volatile void* ptr,
+ [[maybe_unused]] size_t element_size,
+ [[maybe_unused]] size_t num_elements) {
#if RTC_HAS_MSAN
__msan_poison(ptr, element_size * num_elements);
#endif
@@ -83,9 +84,10 @@
// Force an MSan check (if any bits in the memory range [ptr, ptr +
// element_size * num_elements) are uninitialized the call will crash with an
// MSan report).
-static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
- size_t element_size,
- size_t num_elements) {
+static inline void rtc_MsanCheckInitialized(
+ [[maybe_unused]] const volatile void* ptr,
+ [[maybe_unused]] size_t element_size,
+ [[maybe_unused]] size_t num_elements) {
#if RTC_HAS_MSAN
__msan_check_mem_is_initialized(ptr, element_size * num_elements);
#endif
diff --git a/rtc_base/socket.h b/rtc_base/socket.h
index 10e6037..4b9a169 100644
--- a/rtc_base/socket.h
+++ b/rtc_base/socket.h
@@ -113,10 +113,10 @@
// `timestamp` is in units of microseconds.
virtual int Recv(void* pv, size_t cb, int64_t* timestamp) = 0;
// TODO(webrtc:15368): Deprecate and remove.
- virtual int RecvFrom(void* pv,
- size_t cb,
- SocketAddress* paddr,
- int64_t* timestamp) {
+ virtual int RecvFrom(void* /* pv */,
+ size_t /* cb */,
+ SocketAddress* /* paddr */,
+ int64_t* /* timestamp */) {
// Not implemented. Use RecvFrom(ReceiveBuffer& buffer).
RTC_CHECK_NOTREACHED();
}
diff --git a/rtc_base/string_to_number.h b/rtc_base/string_to_number.h
index 5e8aea5..d6e3b38 100644
--- a/rtc_base/string_to_number.h
+++ b/rtc_base/string_to_number.h
@@ -92,7 +92,7 @@
template <typename T>
typename std::enable_if<std::is_floating_point<T>::value,
std::optional<T>>::type
-StringToNumber(absl::string_view str, int base = 10) {
+StringToNumber(absl::string_view str, int /* base */ = 10) {
static_assert(
std::numeric_limits<T>::max() <= std::numeric_limits<long double>::max(),
"StringToNumber only supports floating-point numbers as large "
diff --git a/rtc_base/synchronization/sequence_checker_internal.h b/rtc_base/synchronization/sequence_checker_internal.h
index a23ac08..eb3bf36 100644
--- a/rtc_base/synchronization/sequence_checker_internal.h
+++ b/rtc_base/synchronization/sequence_checker_internal.h
@@ -59,8 +59,8 @@
// right version for your build configuration.
class SequenceCheckerDoNothing {
public:
- explicit SequenceCheckerDoNothing(bool attach_to_current_thread) {}
- explicit SequenceCheckerDoNothing(TaskQueueBase* attached_queue) {}
+ explicit SequenceCheckerDoNothing(bool /* attach_to_current_thread */) {}
+ explicit SequenceCheckerDoNothing(TaskQueueBase* /* attached_queue */) {}
bool IsCurrent() const { return true; }
void Detach() {}
};
@@ -68,7 +68,7 @@
template <typename ThreadLikeObject>
std::enable_if_t<std::is_base_of_v<SequenceCheckerImpl, ThreadLikeObject>,
std::string>
-ExpectationToString(const ThreadLikeObject* checker) {
+ExpectationToString([[maybe_unused]] const ThreadLikeObject* checker) {
#if RTC_DCHECK_IS_ON
return checker->ExpectationToString();
#else
