diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h index f3ca2fa848..a946e2b1f6 100644 --- a/audio/audio_transport_impl.h +++ b/audio/audio_transport_impl.h @@ -42,20 +42,20 @@ class AudioTransportImpl : public AudioTransport { ~AudioTransportImpl() override; int32_t RecordedDataIsAvailable(const void* audioSamples, - const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, - const uint32_t totalDelayMS, - const int32_t clockDrift, - const uint32_t currentMicLevel, - const bool keyPressed, + size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, + uint32_t totalDelayMS, + int32_t clockDrift, + uint32_t currentMicLevel, + bool keyPressed, uint32_t& newMicLevel) override; - int32_t NeedMorePlayData(const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, + int32_t NeedMorePlayData(size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, diff --git a/call/rtp_payload_params.h b/call/rtp_payload_params.h index 48b0ec2c61..1f03fab6da 100644 --- a/call/rtp_payload_params.h +++ b/call/rtp_payload_params.h @@ -32,7 +32,7 @@ class RtpRtcp; // TODO(nisse): Make these properties not codec specific. class RtpPayloadParams final { public: - RtpPayloadParams(const uint32_t ssrc, + RtpPayloadParams(uint32_t ssrc, const RtpPayloadState* state, const WebRtcKeyValueConfig& trials); RtpPayloadParams(const RtpPayloadParams& other); diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h index 378f902d75..7023804506 100644 --- a/call/rtp_video_sender.h +++ b/call/rtp_video_sender.h @@ -96,7 +96,7 @@ class RtpVideoSender : public RtpVideoSenderInterface, void SetActive(bool active) RTC_LOCKS_EXCLUDED(mutex_) override; // Sets the sending status of the rtp modules and appropriately sets the // payload router to active if any rtp modules are active. - void SetActiveModules(const std::vector active_modules) + void SetActiveModules(std::vector active_modules) RTC_LOCKS_EXCLUDED(mutex_) override; bool IsActive() RTC_LOCKS_EXCLUDED(mutex_) override; @@ -153,7 +153,7 @@ class RtpVideoSender : public RtpVideoSenderInterface, private: bool IsActiveLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); - void SetActiveModulesLocked(const std::vector active_modules) + void SetActiveModulesLocked(std::vector active_modules) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); void ConfigureProtection(); diff --git a/call/rtp_video_sender_interface.h b/call/rtp_video_sender_interface.h index a0b4baccb4..acb68e3ae2 100644 --- a/call/rtp_video_sender_interface.h +++ b/call/rtp_video_sender_interface.h @@ -36,7 +36,7 @@ class RtpVideoSenderInterface : public EncodedImageCallback, virtual void SetActive(bool active) = 0; // Sets the sending status of the rtp modules and appropriately sets the // RtpVideoSender to active if any rtp modules are active. - virtual void SetActiveModules(const std::vector active_modules) = 0; + virtual void SetActiveModules(std::vector active_modules) = 0; virtual bool IsActive() = 0; virtual void OnNetworkAvailability(bool network_available) = 0; diff --git a/examples/androidvoip/jni/android_voip_client.h b/examples/androidvoip/jni/android_voip_client.h index bfca7e8b79..8e1edd5ef9 100644 --- a/examples/androidvoip/jni/android_voip_client.h +++ b/examples/androidvoip/jni/android_voip_client.h @@ -147,8 +147,8 @@ class AndroidVoipClient : public webrtc::Transport, // Overloaded methods having native C++ variables as arguments. void SetEncoder(const std::string& encoder); void SetDecoders(const std::vector& decoders); - void SetLocalAddress(const std::string& ip_address, const int port_number); - void SetRemoteAddress(const std::string& ip_address, const int port_number); + void SetLocalAddress(const std::string& ip_address, int port_number); + void SetRemoteAddress(const std::string& ip_address, int port_number); // Methods to send and receive RTP/RTCP packets. Takes in a // copy of a packet as a vector to prolong the lifetime of diff --git a/examples/unityplugin/simple_peer_connection.h b/examples/unityplugin/simple_peer_connection.h index b99bde930e..d5cebc9940 100644 --- a/examples/unityplugin/simple_peer_connection.h +++ b/examples/unityplugin/simple_peer_connection.h @@ -31,7 +31,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver, ~SimplePeerConnection() {} bool InitializePeerConnection(const char** turn_urls, - const int no_of_urls, + int no_of_urls, const char* username, const char* credential, bool is_receiver); @@ -56,13 +56,13 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver, ICECANDIDATEREADYTOSEND_CALLBACK callback); bool SetRemoteDescription(const char* type, const char* sdp); bool AddIceCandidate(const char* sdp, - const int sdp_mlineindex, + int sdp_mlineindex, const char* sdp_mid); protected: // create a peerconneciton and add the turn servers info to the configuration. bool CreatePeerConnection(const char** turn_urls, - const int no_of_urls, + int no_of_urls, const char* username, const char* credential); void CloseDataChannel(); diff --git a/examples/unityplugin/unity_plugin_apis.h b/examples/unityplugin/unity_plugin_apis.h index b32f9e2caf..8b8fe0fe80 100644 --- a/examples/unityplugin/unity_plugin_apis.h +++ b/examples/unityplugin/unity_plugin_apis.h @@ -31,7 +31,7 @@ typedef void (*DATAFROMEDATECHANNELREADY_CALLBACK)(const char* msg); typedef void (*FAILURE_CALLBACK)(const char* msg); typedef void (*LOCALSDPREADYTOSEND_CALLBACK)(const char* type, const char* sdp); typedef void (*ICECANDIDATEREADYTOSEND_CALLBACK)(const char* candidate, - const int sdp_mline_index, + int sdp_mline_index, const char* sdp_mid); typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data, int bits_per_sample, @@ -47,7 +47,7 @@ typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data, extern "C" { // Create a peerconnection and return a unique peer connection id. WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls, - const int no_of_urls, + int no_of_urls, const char* username, const char* credential, bool mandatory_receive_video); @@ -75,9 +75,9 @@ WEBRTC_PLUGIN_API bool SetRemoteDescription(int peer_connection_id, const char* type, const char* sdp); // Add ice candidate. -WEBRTC_PLUGIN_API bool AddIceCandidate(const int peer_connection_id, +WEBRTC_PLUGIN_API bool AddIceCandidate(int peer_connection_id, const char* candidate, - const int sdp_mlineindex, + int sdp_mlineindex, const char* sdp_mid); // Register callback functions. diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h index 398f63de13..6b0270b127 100644 --- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h +++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h @@ -132,7 +132,7 @@ class EventParser { uint64_t ReadOptionalValuePositions(); void ReadDeltasAndPopulateValues(FixedLengthEncodingParametersV3 params, uint64_t num_deltas, - const uint64_t base); + uint64_t base); RtcEventLogParseStatus ParseNumericFieldInternal(uint64_t value_bit_width, FieldType field_type); RtcEventLogParseStatus ParseStringFieldInternal(); diff --git a/media/base/fake_rtp.h b/media/base/fake_rtp.h index f2578151ed..2fa37a1231 100644 --- a/media/base/fake_rtp.h +++ b/media/base/fake_rtp.h @@ -295,7 +295,7 @@ void CompareHeaderExtensions(const char* packet1, size_t packet1_size, const char* packet2, size_t packet2_size, - const std::vector encrypted_headers, + std::vector encrypted_headers, bool expect_equal); #endif // MEDIA_BASE_FAKE_RTP_H_ diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h index cc4ed555cf..d112bfe7f2 100644 --- a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h +++ b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h @@ -38,7 +38,7 @@ int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc* streamData, int16_t* dataQ7, const uint16_t* env, - const int16_t lenData); + int16_t lenData); /**************************************************************************** * WebRtcIsacfix_EncTerminate(...) @@ -73,7 +73,7 @@ int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc* streamData); int WebRtcIsacfix_DecLogisticMulti2(int16_t* data, Bitstr_dec* streamData, const int32_t* env, - const int16_t lenData); + int16_t lenData); /**************************************************************************** * WebRtcIsacfix_EncHistMulti(...) @@ -92,7 +92,7 @@ int WebRtcIsacfix_DecLogisticMulti2(int16_t* data, int WebRtcIsacfix_EncHistMulti(Bitstr_enc* streamData, const int16_t* data, const uint16_t* const* cdf, - const int16_t lenData); + int16_t lenData); /**************************************************************************** * WebRtcIsacfix_DecHistBisectMulti(...) @@ -118,7 +118,7 @@ int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t* data, Bitstr_dec* streamData, const uint16_t* const* cdf, const uint16_t* cdfSize, - const int16_t lenData); + int16_t lenData); /**************************************************************************** * WebRtcIsacfix_DecHistOneStepMulti(...) @@ -144,6 +144,6 @@ int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t* data, Bitstr_dec* streamData, const uint16_t* const* cdf, const uint16_t* initIndex, - const int16_t lenData); + int16_t lenData); #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h index ebb74d6c49..f106746f14 100644 --- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h +++ b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h @@ -56,17 +56,17 @@ int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr* bwest_str); */ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr* bwest_str, - const uint16_t rtp_number, - const int16_t frameSize, - const uint32_t send_ts, - const uint32_t arr_ts, - const size_t pksize, - const uint16_t Index); + uint16_t rtp_number, + int16_t frameSize, + uint32_t send_ts, + uint32_t arr_ts, + size_t pksize, + uint16_t Index); /* Update receiving estimates. Used when we only receive BWE index, no iSAC data * packet. */ int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr* bwest_str, - const int16_t Index); + int16_t Index); /**************************************************************************** * WebRtcIsacfix_GetDownlinkBwIndexImpl(...) @@ -100,19 +100,19 @@ int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str); */ uint16_t WebRtcIsacfix_GetMinBytes( RateModel* State, - int16_t StreamSize, /* bytes in bitstream */ - const int16_t FrameLen, /* ms per frame */ - const int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */ - const int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */ + int16_t StreamSize, /* bytes in bitstream */ + int16_t FrameLen, /* ms per frame */ + int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */ + int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */ /* * update long-term average bitrate and amount of data in buffer */ void WebRtcIsacfix_UpdateRateModel( RateModel* State, - int16_t StreamSize, /* bytes in bitstream */ - const int16_t FrameSamples, /* samples per frame */ - const int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */ + int16_t StreamSize, /* bytes in bitstream */ + int16_t FrameSamples, /* samples per frame */ + int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */ void WebRtcIsacfix_InitRateModel(RateModel* State); diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h index b4251cee1e..ae11394f7c 100644 --- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h +++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h @@ -101,19 +101,19 @@ void WebRtcIsacfix_TranscodeLpcCoef(int32_t* tmpcoeffs_gQ6, int16_t* index_gQQ); typedef void (*MatrixProduct1)(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift); + int matrix1_index_factor1, + int matrix0_index_factor1, + int matrix1_index_init_case, + int matrix1_index_step, + int matrix0_index_step, + int inner_loop_count, + int mid_loop_count, + int shift); typedef void (*MatrixProduct2)(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step); + int matrix0_index_factor, + int matrix0_index_step); extern MatrixProduct1 WebRtcIsacfix_MatrixProduct1; extern MatrixProduct2 WebRtcIsacfix_MatrixProduct2; @@ -121,57 +121,57 @@ extern MatrixProduct2 WebRtcIsacfix_MatrixProduct2; void WebRtcIsacfix_MatrixProduct1C(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift); + int matrix1_index_factor1, + int matrix0_index_factor1, + int matrix1_index_init_case, + int matrix1_index_step, + int matrix0_index_step, + int inner_loop_count, + int mid_loop_count, + int shift); void WebRtcIsacfix_MatrixProduct2C(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step); + int matrix0_index_factor, + int matrix0_index_step); #if defined(WEBRTC_HAS_NEON) void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift); + int matrix1_index_factor1, + int matrix0_index_factor1, + int matrix1_index_init_case, + int matrix1_index_step, + int matrix0_index_step, + int inner_loop_count, + int mid_loop_count, + int shift); void WebRtcIsacfix_MatrixProduct2Neon(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step); + int matrix0_index_factor, + int matrix0_index_step); #endif #if defined(MIPS32_LE) void WebRtcIsacfix_MatrixProduct1MIPS(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift); + int matrix1_index_factor1, + int matrix0_index_factor1, + int matrix1_index_init_case, + int matrix1_index_step, + int matrix0_index_step, + int inner_loop_count, + int mid_loop_count, + int shift); void WebRtcIsacfix_MatrixProduct2MIPS(const int16_t matrix0[], const int32_t matrix1[], int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step); + int matrix0_index_factor, + int matrix0_index_step); #endif #endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h index 6b99914b64..f741e6f677 100644 --- a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h +++ b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h @@ -46,7 +46,7 @@ typedef void (*AllpassFilter2FixDec16)( int16_t* data_ch2, // Input and output in channel 2, in Q0 const int16_t* factor_ch1, // Scaling factor for channel 1, in Q15 const int16_t* factor_ch2, // Scaling factor for channel 2, in Q15 - const int length, // Length of the data buffers + int length, // Length of the data buffers int32_t* filter_state_ch1, // Filter state for channel 1, in Q16 int32_t* filter_state_ch2); // Filter state for channel 2, in Q16 extern AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16; @@ -55,7 +55,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C(int16_t* data_ch1, int16_t* data_ch2, const int16_t* factor_ch1, const int16_t* factor_ch2, - const int length, + int length, int32_t* filter_state_ch1, int32_t* filter_state_ch2); @@ -64,7 +64,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16Neon(int16_t* data_ch1, int16_t* data_ch2, const int16_t* factor_ch1, const int16_t* factor_ch2, - const int length, + int length, int32_t* filter_state_ch1, int32_t* filter_state_ch2); #endif @@ -74,7 +74,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16MIPS(int16_t* data_ch1, int16_t* data_ch2, const int16_t* factor_ch1, const int16_t* factor_ch2, - const int length, + int length, int32_t* filter_state_ch1, int32_t* filter_state_ch2); #endif diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h index 221e65ff3f..5f4550a3a5 100644 --- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h +++ b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h @@ -92,11 +92,11 @@ int32_t WebRtcIsac_InitBandwidthEstimator( * estimated by other side */ /* returns 0 if everything went fine, -1 otherwise */ int16_t WebRtcIsac_UpdateBandwidthEstimator(BwEstimatorstr* bwest_str, - const uint16_t rtp_number, - const int32_t frame_length, - const uint32_t send_ts, - const uint32_t arr_ts, - const size_t pksize); + uint16_t rtp_number, + int32_t frame_length, + uint32_t send_ts, + uint32_t arr_ts, + size_t pksize); /* Update receiving estimates. Used when we only receive BWE index, no iSAC data * packet. */ @@ -131,10 +131,10 @@ int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str); */ int WebRtcIsac_GetMinBytes( RateModel* State, - int StreamSize, /* bytes in bitstream */ - const int FrameLen, /* ms per frame */ - const double BottleNeck, /* bottle neck rate; excl headers (bps) */ - const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */ + int StreamSize, /* bytes in bitstream */ + int FrameLen, /* ms per frame */ + double BottleNeck, /* bottle neck rate; excl headers (bps) */ + double DelayBuildUp, /* max delay from bottleneck buffering (ms) */ enum ISACBandwidth bandwidth /*,int16_t frequentLargePackets*/); @@ -143,9 +143,9 @@ int WebRtcIsac_GetMinBytes( */ void WebRtcIsac_UpdateRateModel( RateModel* State, - int StreamSize, /* bytes in bitstream */ - const int FrameSamples, /* samples per frame */ - const double BottleNeck); /* bottle neck rate; excl headers (bps) */ + int StreamSize, /* bytes in bitstream */ + int FrameSamples, /* samples per frame */ + double BottleNeck); /* bottle neck rate; excl headers (bps) */ void WebRtcIsac_InitRateModel(RateModel* State); diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h index 003d966fbd..8b518fb979 100644 --- a/modules/audio_coding/include/audio_coding_module.h +++ b/modules/audio_coding/include/audio_coding_module.h @@ -190,7 +190,7 @@ class AudioCodingModule { // 0 if payload is successfully pushed in. // virtual int32_t IncomingPacket(const uint8_t* incoming_payload, - const size_t payload_len_bytes, + size_t payload_len_bytes, const RTPHeader& rtp_header) = 0; /////////////////////////////////////////////////////////////////////////// diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h index a73be2dd68..076960a21c 100644 --- a/modules/audio_coding/neteq/tools/audio_loop.h +++ b/modules/audio_coding/neteq/tools/audio_loop.h @@ -36,7 +36,7 @@ class AudioLoop { // greater. Otherwise, the loop length is the same as the file length. // The audio will be delivered in blocks of `block_length_samples`. // Returns false if the initialization failed, otherwise true. - bool Init(const std::string file_name, + bool Init(std::string file_name, size_t max_loop_length_samples, size_t block_length_samples); diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h index c6e65a0c35..056dbf5c20 100644 --- a/modules/audio_coding/neteq/tools/input_audio_file.h +++ b/modules/audio_coding/neteq/tools/input_audio_file.h @@ -21,7 +21,7 @@ namespace test { // Class for handling a looping input audio file. class InputAudioFile { public: - explicit InputAudioFile(const std::string file_name, bool loop_at_end = true); + explicit InputAudioFile(std::string file_name, bool loop_at_end = true); virtual ~InputAudioFile(); diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h index c96a4d69f4..fbc42fbe0e 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.h +++ b/modules/audio_coding/test/EncodeDecodeTest.h @@ -28,11 +28,11 @@ class TestPacketization : public AudioPacketizationCallback { public: TestPacketization(RTPStream* rtpStream, uint16_t frequency); ~TestPacketization(); - int32_t SendData(const AudioFrameType frameType, - const uint8_t payloadType, - const uint32_t timeStamp, + int32_t SendData(AudioFrameType frameType, + uint8_t payloadType, + uint32_t timeStamp, const uint8_t* payloadData, - const size_t payloadSize, + size_t payloadSize, int64_t absolute_capture_timestamp_ms) override; private: diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h index a3d1520922..a9f574bab0 100644 --- a/modules/audio_coding/test/RTPFile.h +++ b/modules/audio_coding/test/RTPFile.h @@ -25,11 +25,11 @@ class RTPStream { public: virtual ~RTPStream() {} - virtual void Write(const uint8_t payloadType, - const uint32_t timeStamp, - const int16_t seqNo, + virtual void Write(uint8_t payloadType, + uint32_t timeStamp, + int16_t seqNo, const uint8_t* payloadData, - const size_t payloadSize, + size_t payloadSize, uint32_t frequency) = 0; // Returns the packet's payload size. Zero should be treated as an @@ -75,11 +75,11 @@ class RTPBuffer : public RTPStream { ~RTPBuffer() = default; - void Write(const uint8_t payloadType, - const uint32_t timeStamp, - const int16_t seqNo, + void Write(uint8_t payloadType, + uint32_t timeStamp, + int16_t seqNo, const uint8_t* payloadData, - const size_t payloadSize, + size_t payloadSize, uint32_t frequency) override; size_t Read(RTPHeader* rtp_header, @@ -108,11 +108,11 @@ class RTPFile : public RTPStream { void ReadHeader(); - void Write(const uint8_t payloadType, - const uint32_t timeStamp, - const int16_t seqNo, + void Write(uint8_t payloadType, + uint32_t timeStamp, + int16_t seqNo, const uint8_t* payloadData, - const size_t payloadSize, + size_t payloadSize, uint32_t frequency) override; size_t Read(RTPHeader* rtp_header, diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h index 3ee4dbf594..4c50a4b555 100644 --- a/modules/audio_coding/test/TestStereo.h +++ b/modules/audio_coding/test/TestStereo.h @@ -31,11 +31,11 @@ class TestPackStereo : public AudioPacketizationCallback { void RegisterReceiverACM(AudioCodingModule* acm); - int32_t SendData(const AudioFrameType frame_type, - const uint8_t payload_type, - const uint32_t timestamp, + int32_t SendData(AudioFrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, const uint8_t* payload_data, - const size_t payload_size, + size_t payload_size, int64_t absolute_capture_timestamp_ms) override; uint16_t payload_size(); diff --git a/modules/audio_coding/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h index 7d0cdb9566..b7eb9e5583 100644 --- a/modules/audio_coding/test/TwoWayCommunication.h +++ b/modules/audio_coding/test/TwoWayCommunication.h @@ -31,9 +31,9 @@ class TwoWayCommunication { private: void SetUpAutotest(AudioEncoderFactory* const encoder_factory, const SdpAudioFormat& format1, - const int payload_type1, + int payload_type1, const SdpAudioFormat& format2, - const int payload_type2); + int payload_type2); std::unique_ptr _acmA; std::unique_ptr _acmB; diff --git a/modules/audio_device/include/audio_device_data_observer.h b/modules/audio_device/include/audio_device_data_observer.h index b59cafcb5d..0e93202125 100644 --- a/modules/audio_device/include/audio_device_data_observer.h +++ b/modules/audio_device/include/audio_device_data_observer.h @@ -26,16 +26,16 @@ namespace webrtc { class AudioDeviceDataObserver { public: virtual void OnCaptureData(const void* audio_samples, - const size_t num_samples, - const size_t bytes_per_sample, - const size_t num_channels, + size_t num_samples, + size_t bytes_per_sample, + size_t num_channels, const uint32_t samples_per_sec) = 0; virtual void OnRenderData(const void* audio_samples, - const size_t num_samples, - const size_t bytes_per_sample, - const size_t num_channels, - const uint32_t samples_per_sec) = 0; + size_t num_samples, + size_t bytes_per_sample, + size_t num_channels, + uint32_t samples_per_sec) = 0; AudioDeviceDataObserver() = default; virtual ~AudioDeviceDataObserver() = default; @@ -56,14 +56,14 @@ rtc::scoped_refptr CreateAudioDeviceWithDataObserver( // Creates an ADM instance with AudioDeviceDataObserver registered. rtc::scoped_refptr CreateAudioDeviceWithDataObserver( - const AudioDeviceModule::AudioLayer audio_layer, + AudioDeviceModule::AudioLayer audio_layer, TaskQueueFactory* task_queue_factory, std::unique_ptr observer); // Creates an ADM instance with AudioDeviceDataObserver registered. ABSL_DEPRECATED("") rtc::scoped_refptr CreateAudioDeviceWithDataObserver( - const AudioDeviceModule::AudioLayer audio_layer, + AudioDeviceModule::AudioLayer audio_layer, TaskQueueFactory* task_queue_factory, AudioDeviceDataObserver* observer); diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h index 01129a47a9..7b2c784c30 100644 --- a/modules/audio_device/include/audio_device_defines.h +++ b/modules/audio_device/include/audio_device_defines.h @@ -34,21 +34,21 @@ static const int kAdmMaxPlayoutBufferSizeMs = 250; class AudioTransport { public: virtual int32_t RecordedDataIsAvailable(const void* audioSamples, - const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, - const uint32_t totalDelayMS, - const int32_t clockDrift, - const uint32_t currentMicLevel, - const bool keyPressed, + size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, + uint32_t totalDelayMS, + int32_t clockDrift, + uint32_t currentMicLevel, + bool keyPressed, uint32_t& newMicLevel) = 0; // NOLINT // Implementation has to setup safe values for all specified out parameters. - virtual int32_t NeedMorePlayData(const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, + virtual int32_t NeedMorePlayData(size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, // NOLINT int64_t* elapsed_time_ms, diff --git a/modules/audio_device/include/mock_audio_transport.h b/modules/audio_device/include/mock_audio_transport.h index 8f71a2d71f..bcba33865a 100644 --- a/modules/audio_device/include/mock_audio_transport.h +++ b/modules/audio_device/include/mock_audio_transport.h @@ -25,23 +25,23 @@ class MockAudioTransport : public AudioTransport { MOCK_METHOD(int32_t, RecordedDataIsAvailable, (const void* audioSamples, - const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, - const uint32_t totalDelayMS, - const int32_t clockDrift, - const uint32_t currentMicLevel, - const bool keyPressed, + size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, + uint32_t totalDelayMS, + int32_t clockDrift, + uint32_t currentMicLevel, + bool keyPressed, uint32_t& newMicLevel), (override)); MOCK_METHOD(int32_t, NeedMorePlayData, - (const size_t nSamples, - const size_t nBytesPerSample, - const size_t nChannels, - const uint32_t samplesPerSec, + (size_t nSamples, + size_t nBytesPerSample, + size_t nChannels, + uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, diff --git a/modules/audio_device/linux/audio_device_alsa_linux.h b/modules/audio_device/linux/audio_device_alsa_linux.h index 1f4a231640..23e21d3ce9 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.h +++ b/modules/audio_device/linux/audio_device_alsa_linux.h @@ -131,11 +131,11 @@ class AudioDeviceLinuxALSA : public AudioDeviceGeneric { int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); - int32_t GetDevicesInfo(const int32_t function, - const bool playback, - const int32_t enumDeviceNo = 0, + int32_t GetDevicesInfo(int32_t function, + bool playback, + int32_t enumDeviceNo = 0, char* enumDeviceName = NULL, - const int32_t ednLen = 0) const; + int32_t ednLen = 0) const; int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle); bool KeyPressed() const; diff --git a/modules/audio_device/mac/audio_device_mac.h b/modules/audio_device/mac/audio_device_mac.h index 5ed3d479ee..fe0d3beed8 100644 --- a/modules/audio_device/mac/audio_device_mac.h +++ b/modules/audio_device/mac/audio_device_mac.h @@ -169,16 +169,16 @@ class AudioDeviceMac : public AudioDeviceGeneric { static void AtomicSet32(int32_t* theValue, int32_t newValue); static int32_t AtomicGet32(int32_t* theValue); - static void logCAMsg(const rtc::LoggingSeverity sev, + static void logCAMsg(rtc::LoggingSeverity sev, const char* msg, const char* err); - int32_t GetNumberDevices(const AudioObjectPropertyScope scope, + int32_t GetNumberDevices(AudioObjectPropertyScope scope, AudioDeviceID scopedDeviceIds[], - const uint32_t deviceListLength); + uint32_t deviceListLength); - int32_t GetDeviceName(const AudioObjectPropertyScope scope, - const uint16_t index, + int32_t GetDeviceName(AudioObjectPropertyScope scope, + uint16_t index, char* name); int32_t InitDevice(uint16_t userDeviceIndex, diff --git a/modules/audio_device/mac/audio_mixer_manager_mac.h b/modules/audio_device/mac/audio_mixer_manager_mac.h index 17e34ff17d..0ccab4879b 100644 --- a/modules/audio_device/mac/audio_mixer_manager_mac.h +++ b/modules/audio_device/mac/audio_mixer_manager_mac.h @@ -54,7 +54,7 @@ class AudioMixerManagerMac { private: int32_t CloseSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); int32_t CloseMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); - static void logCAMsg(const rtc::LoggingSeverity sev, + static void logCAMsg(rtc::LoggingSeverity sev, const char* msg, const char* err); diff --git a/modules/audio_device/win/core_audio_utility_win.h b/modules/audio_device/win/core_audio_utility_win.h index fa7b198bb1..754b4ddafa 100644 --- a/modules/audio_device/win/core_audio_utility_win.h +++ b/modules/audio_device/win/core_audio_utility_win.h @@ -537,7 +537,7 @@ bool FillRenderEndpointBufferWithSilence(IAudioClient* client, // Prints/logs all fields of the format structure in `format`. // Also supports extended versions (WAVEFORMATEXTENSIBLE). -std::string WaveFormatToString(const WaveFormatWrapper format); +std::string WaveFormatToString(WaveFormatWrapper format); // Converts Windows internal REFERENCE_TIME (100 nanosecond units) into // generic webrtc::TimeDelta which then can be converted to any time unit. diff --git a/modules/audio_processing/aec3/aec3_common.h b/modules/audio_processing/aec3/aec3_common.h index 3bfff967a0..32b564f14b 100644 --- a/modules/audio_processing/aec3/aec3_common.h +++ b/modules/audio_processing/aec3/aec3_common.h @@ -85,10 +85,10 @@ constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor, Aec3Optimization DetectOptimization(); // Computes the log2 of the input in a fast an approximate manner. -float FastApproxLog2f(const float in); +float FastApproxLog2f(float in); // Returns dB from a power quantity expressed in log2. -float Log2TodB(const float in_log2); +float Log2TodB(float in_log2); static_assert(1 << kBlockSizeLog2 == kBlockSize, "Proper number of shifts for blocksize"); diff --git a/modules/audio_processing/aec3/fullband_erle_estimator.h b/modules/audio_processing/aec3/fullband_erle_estimator.h index 2b720a4de4..7a082176d6 100644 --- a/modules/audio_processing/aec3/fullband_erle_estimator.h +++ b/modules/audio_processing/aec3/fullband_erle_estimator.h @@ -67,7 +67,7 @@ class FullBandErleEstimator { // Updates the estimator with a new point, returns true // if the instantaneous ERLE was updated due to having enough // points for performing the estimate. - bool Update(const float Y2_sum, const float E2_sum); + bool Update(float Y2_sum, float E2_sum); // Resets the instantaneous ERLE estimator to its initial state. void Reset(); // Resets the members related with an instantaneous estimate. diff --git a/modules/audio_processing/aecm/aecm_core.h b/modules/audio_processing/aecm/aecm_core.h index d6d0d8dafc..2ee1129daf 100644 --- a/modules/audio_processing/aecm/aecm_core.h +++ b/modules/audio_processing/aecm/aecm_core.h @@ -248,7 +248,7 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm, // void WebRtcAecm_BufferFarFrame(AecmCore* const aecm, const int16_t* const farend, - const int farLen); + int farLen); //////////////////////////////////////////////////////////////////////////////// // WebRtcAecm_FetchFarFrame() @@ -263,8 +263,8 @@ void WebRtcAecm_BufferFarFrame(AecmCore* const aecm, // void WebRtcAecm_FetchFarFrame(AecmCore* const aecm, int16_t* const farend, - const int farLen, - const int knownDelay); + int farLen, + int knownDelay); // All the functions below are intended to be private @@ -374,9 +374,9 @@ int16_t WebRtcAecm_CalcStepSize(AecmCore* const aecm); // void WebRtcAecm_UpdateChannel(AecmCore* aecm, const uint16_t* far_spectrum, - const int16_t far_q, + int16_t far_q, const uint16_t* const dfa, - const int16_t mu, + int16_t mu, int32_t* echoEst); extern const int16_t WebRtcAecm_kCosTable[]; diff --git a/modules/audio_processing/test/fake_recording_device.h b/modules/audio_processing/test/fake_recording_device.h index 4017037137..da3c0cf794 100644 --- a/modules/audio_processing/test/fake_recording_device.h +++ b/modules/audio_processing/test/fake_recording_device.h @@ -48,8 +48,8 @@ class FakeRecordingDevice final { ~FakeRecordingDevice(); int MicLevel() const; - void SetMicLevel(const int level); - void SetUndoMicLevel(const int level); + void SetMicLevel(int level); + void SetUndoMicLevel(int level); // Simulates the analog gain. // If `real_device_level` is a valid level, the unmodified mic signal is diff --git a/modules/congestion_controller/goog_cc/test/goog_cc_printer.h b/modules/congestion_controller/goog_cc/test/goog_cc_printer.h index 3eee7814cf..16fa657e71 100644 --- a/modules/congestion_controller/goog_cc/test/goog_cc_printer.h +++ b/modules/congestion_controller/goog_cc/test/goog_cc_printer.h @@ -61,7 +61,7 @@ class GoogCcDebugFactory : public GoogCcNetworkControllerFactory { std::unique_ptr Create( NetworkControllerConfig config) override; - void PrintState(const Timestamp at_time); + void PrintState(Timestamp at_time); void AttachWriter(std::unique_ptr log_writer); diff --git a/modules/desktop_capture/win/screen_capture_utils.h b/modules/desktop_capture/win/screen_capture_utils.h index ac60c5abc5..bcb183b9d2 100644 --- a/modules/desktop_capture/win/screen_capture_utils.h +++ b/modules/desktop_capture/win/screen_capture_utils.h @@ -39,24 +39,23 @@ bool GetScreenList(DesktopCapturer::SourceList* screens, // Converts a device index (which are returned by `GetScreenList`) into an // HMONITOR. -bool GetHmonitorFromDeviceIndex(const DesktopCapturer::SourceId device_index, +bool GetHmonitorFromDeviceIndex(DesktopCapturer::SourceId device_index, HMONITOR* hmonitor); // Returns true if `monitor` represents a valid display // monitor. Consumers should recheck the validity of HMONITORs before use if a // WM_DISPLAYCHANGE message has been received. -bool IsMonitorValid(const HMONITOR monitor); +bool IsMonitorValid(HMONITOR monitor); // Returns the rect of the monitor identified by `monitor`, relative to the // primary display's top-left. On failure, returns an empty rect. -DesktopRect GetMonitorRect(const HMONITOR monitor); +DesktopRect GetMonitorRect(HMONITOR monitor); // Returns true if `screen` is a valid screen. The screen device key is // returned through `device_key` if the screen is valid. The device key can be // used in GetScreenRect to verify the screen matches the previously obtained // id. -bool IsScreenValid(const DesktopCapturer::SourceId screen, - std::wstring* device_key); +bool IsScreenValid(DesktopCapturer::SourceId screen, std::wstring* device_key); // Get the rect of the entire system in system coordinate system. I.e. the // primary monitor always starts from (0, 0). @@ -65,7 +64,7 @@ DesktopRect GetFullscreenRect(); // Get the rect of the screen identified by `screen`, relative to the primary // display's top-left. If the screen device key does not match `device_key`, or // the screen does not exist, or any error happens, an empty rect is returned. -RTC_EXPORT DesktopRect GetScreenRect(const DesktopCapturer::SourceId screen, +RTC_EXPORT DesktopRect GetScreenRect(DesktopCapturer::SourceId screen, const std::wstring& device_key); } // namespace webrtc diff --git a/modules/desktop_capture/win/test_support/test_window.h b/modules/desktop_capture/win/test_support/test_window.h index 8701dc990b..b055da7ccd 100644 --- a/modules/desktop_capture/win/test_support/test_window.h +++ b/modules/desktop_capture/win/test_support/test_window.h @@ -32,17 +32,17 @@ struct WindowInfo { }; WindowInfo CreateTestWindow(const WCHAR* window_title, - const int height = 0, - const int width = 0, - const LONG extended_styles = 0); + int height = 0, + int width = 0, + LONG extended_styles = 0); -void ResizeTestWindow(const HWND hwnd, const int width, const int height); +void ResizeTestWindow(HWND hwnd, int width, int height); -void MoveTestWindow(const HWND hwnd, const int x, const int y); +void MoveTestWindow(HWND hwnd, int x, int y); -void MinimizeTestWindow(const HWND hwnd); +void MinimizeTestWindow(HWND hwnd); -void UnminimizeTestWindow(const HWND hwnd); +void UnminimizeTestWindow(HWND hwnd); void DestroyTestWindow(WindowInfo info); diff --git a/modules/video_capture/device_info_impl.h b/modules/video_capture/device_info_impl.h index 4b47389609..546265049c 100644 --- a/modules/video_capture/device_info_impl.h +++ b/modules/video_capture/device_info_impl.h @@ -29,7 +29,7 @@ class DeviceInfoImpl : public VideoCaptureModule::DeviceInfo { ~DeviceInfoImpl(void) override; int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8) override; int32_t GetCapability(const char* deviceUniqueIdUTF8, - const uint32_t deviceCapabilityNumber, + uint32_t deviceCapabilityNumber, VideoCaptureCapability& capability) override; int32_t GetBestMatchedCapability(const char* deviceUniqueIdUTF8, diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h index 0f60092d72..3bbe217cba 100644 --- a/modules/video_capture/video_capture.h +++ b/modules/video_capture/video_capture.h @@ -44,7 +44,7 @@ class VideoCaptureModule : public rtc::RefCountInterface { // Gets the capabilities of the named device. virtual int32_t GetCapability(const char* deviceUniqueIdUTF8, - const uint32_t deviceCapabilityNumber, + uint32_t deviceCapabilityNumber, VideoCaptureCapability& capability) = 0; // Gets clockwise angle the captured frames should be rotated in order diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h index 93b2a59139..b5e9cc6d4b 100644 --- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h +++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h @@ -69,11 +69,10 @@ class LibvpxVp9Encoder : public VP9Encoder { absl::optional* spatial_idx, const vpx_codec_cx_pkt& pkt); void FillReferenceIndices(const vpx_codec_cx_pkt& pkt, - const size_t pic_num, - const bool inter_layer_predicted, + size_t pic_num, + bool inter_layer_predicted, CodecSpecificInfoVP9* vp9_info); - void UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt, - const size_t pic_num); + void UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt, size_t pic_num); vpx_svc_ref_frame_config_t SetReferences( bool is_key_pic, size_t first_active_spatial_layer_id); @@ -107,7 +106,7 @@ class LibvpxVp9Encoder : public VP9Encoder { size_t SteadyStateSize(int sid, int tid); - void MaybeRewrapRawWithFormat(const vpx_img_fmt fmt); + void MaybeRewrapRawWithFormat(vpx_img_fmt fmt); // Prepares `raw_` to reference image data of `buffer`, or of mapped or scaled // versions of `buffer`. Returns the buffer that got referenced as a result, // allowing the caller to keep a reference to it until after encoding has diff --git a/modules/video_coding/utility/simulcast_test_fixture_impl.h b/modules/video_coding/utility/simulcast_test_fixture_impl.h index a3d3fc66a8..cdfdc609d5 100644 --- a/modules/video_coding/utility/simulcast_test_fixture_impl.h +++ b/modules/video_coding/utility/simulcast_test_fixture_impl.h @@ -64,10 +64,10 @@ class SimulcastTestFixtureImpl final : public SimulcastTestFixture { void SetUpCodec(const int* temporal_layer_profile); void SetUpRateAllocator(); void SetRates(uint32_t bitrate_kbps, uint32_t fps); - void RunActiveStreamsTest(const std::vector active_streams); - void UpdateActiveStreams(const std::vector active_streams); + void RunActiveStreamsTest(std::vector active_streams); + void UpdateActiveStreams(std::vector active_streams); void ExpectStreams(VideoFrameType frame_type, - const std::vector expected_streams_active); + std::vector expected_streams_active); void ExpectStreams(VideoFrameType frame_type, int expected_video_streams); void VerifyTemporalIdxAndSyncForAllSpatialLayers( TestEncodedImageCallback* encoder_callback, diff --git a/modules/video_processing/util/skin_detection.h b/modules/video_processing/util/skin_detection.h index 7be791f2d8..7f2e17aa87 100644 --- a/modules/video_processing/util/skin_detection.h +++ b/modules/video_processing/util/skin_detection.h @@ -19,11 +19,11 @@ typedef unsigned char uint8_t; bool MbHasSkinColor(const uint8_t* y_src, const uint8_t* u_src, const uint8_t* v_src, - const int stride_y, - const int stride_u, - const int stride_v, - const int mb_row, - const int mb_col); + int stride_y, + int stride_u, + int stride_v, + int mb_row, + int mb_col); } // namespace webrtc diff --git a/net/dcsctp/rx/traditional_reassembly_streams.h b/net/dcsctp/rx/traditional_reassembly_streams.h index 0c724327e2..2fac9ff683 100644 --- a/net/dcsctp/rx/traditional_reassembly_streams.h +++ b/net/dcsctp/rx/traditional_reassembly_streams.h @@ -55,8 +55,7 @@ class TraditionalReassemblyStreams : public ReassemblyStreams { explicit StreamBase(TraditionalReassemblyStreams* parent) : parent_(*parent) {} - size_t AssembleMessage(const ChunkMap::iterator start, - const ChunkMap::iterator end); + size_t AssembleMessage(ChunkMap::iterator start, ChunkMap::iterator end); TraditionalReassemblyStreams& parent_; }; diff --git a/net/dcsctp/testing/data_generator.h b/net/dcsctp/testing/data_generator.h index 859450b1c3..f917c740a7 100644 --- a/net/dcsctp/testing/data_generator.h +++ b/net/dcsctp/testing/data_generator.h @@ -38,14 +38,14 @@ class DataGenerator { // "is_end" flag. Data Ordered(std::vector payload, absl::string_view flags = "", - const DataGeneratorOptions opts = {}); + DataGeneratorOptions opts = {}); // Generates unordered "data" with the provided `payload` and flags, which can // contain "B" for setting the "is_beginning" flag, and/or "E" for setting the // "is_end" flag. Data Unordered(std::vector payload, absl::string_view flags = "", - const DataGeneratorOptions opts = {}); + DataGeneratorOptions opts = {}); // Resets the Message ID identifier - simulating a "stream reset". void ResetStream() { message_id_ = MID(0); } diff --git a/p2p/stunprober/stun_prober.h b/p2p/stunprober/stun_prober.h index b562394fcc..b1acd7704d 100644 --- a/p2p/stunprober/stun_prober.h +++ b/p2p/stunprober/stun_prober.h @@ -121,7 +121,7 @@ class RTC_EXPORT StunProber : public sigslot::has_slots<> { int stun_ta_interval_ms, int requests_per_ip, int timeout_ms, - const AsyncCallback finish_callback); + AsyncCallback finish_callback); // TODO(guoweis): The combination of Prepare() and Run() are equivalent to the // Start() above. Remove Start() once everything is migrated. diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h index 4e06566806..d207269d09 100644 --- a/pc/jsep_transport_controller.h +++ b/pc/jsep_transport_controller.h @@ -135,7 +135,7 @@ class JsepTransportController : public sigslot::has_slots<> { // Factory for SCTP transports. SctpTransportFactoryInterface* sctp_factory = nullptr; - std::function on_dtls_handshake_error_; + std::function on_dtls_handshake_error_; }; // The ICE related events are fired on the `network_thread`. diff --git a/pc/rtp_transmission_manager.h b/pc/rtp_transmission_manager.h index f616d9d0f6..bff9b48267 100644 --- a/pc/rtp_transmission_manager.h +++ b/pc/rtp_transmission_manager.h @@ -184,7 +184,7 @@ class RtpTransmissionManager : public RtpSenderBase::SetStreamsObserver { cricket::MediaType media_type); const RtpSenderInfo* FindSenderInfo(const std::vector& infos, const std::string& stream_id, - const std::string sender_id) const; + std::string sender_id) const; // Return the RtpSender with the given track attached. rtc::scoped_refptr> diff --git a/rtc_base/experiments/struct_parameters_parser.h b/rtc_base/experiments/struct_parameters_parser.h index 523ecfb05d..f5f8340209 100644 --- a/rtc_base/experiments/struct_parameters_parser.h +++ b/rtc_base/experiments/struct_parameters_parser.h @@ -28,7 +28,7 @@ namespace webrtc { namespace struct_parser_impl { struct TypedMemberParser { public: - bool (*parse)(const absl::string_view src, void* target); + bool (*parse)(absl::string_view src, void* target); void (*encode)(const void* src, std::string* target); }; diff --git a/rtc_base/string_encode.h b/rtc_base/string_encode.h index d636f7f227..c63d5271fa 100644 --- a/rtc_base/string_encode.h +++ b/rtc_base/string_encode.h @@ -80,7 +80,7 @@ size_t tokenize(absl::string_view source, // duplicates of delimiter ignored. Return false if the delimiter could not be // found, otherwise return true. bool tokenize_first(absl::string_view source, - const char delimiter, + char delimiter, std::string* token, std::string* rest); diff --git a/rtc_base/string_utils.h b/rtc_base/string_utils.h index d844e5e125..6e8b0b599b 100644 --- a/rtc_base/string_utils.h +++ b/rtc_base/string_utils.h @@ -86,7 +86,7 @@ inline std::string ToUtf8(const std::wstring& wstr) { std::string string_trim(const std::string& s); // TODO(jonasolsson): replace with absl::Hex when that becomes available. -std::string ToHex(const int i); +std::string ToHex(int i); // CompileTimeString comprises of a string-like object which can be used as a // regular const char* in compile time and supports concatenation. Useful for diff --git a/sdk/objc/components/video_codec/helpers.h b/sdk/objc/components/video_codec/helpers.h index 0683ea79e5..7c9ef1cd87 100644 --- a/sdk/objc/components/video_codec/helpers.h +++ b/sdk/objc/components/video_codec/helpers.h @@ -26,7 +26,7 @@ inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys, } // Copies characters from a CFStringRef into a std::string. -std::string CFStringToString(const CFStringRef cf_string); +std::string CFStringToString(CFStringRef cf_string); // Convenience function for setting a VT property. void SetVTSessionProperty(VTSessionRef session, CFStringRef key, int32_t value); diff --git a/test/direct_transport.h b/test/direct_transport.h index 34b68555d5..d6b70b9ce0 100644 --- a/test/direct_transport.h +++ b/test/direct_transport.h @@ -32,7 +32,7 @@ class Demuxer { explicit Demuxer(const std::map& payload_type_map); ~Demuxer() = default; MediaType GetMediaType(const uint8_t* packet_data, - const size_t packet_length) const; + size_t packet_length) const; const std::map payload_type_map_; RTC_DISALLOW_COPY_AND_ASSIGN(Demuxer); }; diff --git a/test/pc/e2e/network_quality_metrics_reporter.h b/test/pc/e2e/network_quality_metrics_reporter.h index 50c36234a5..5cedce1e7f 100644 --- a/test/pc/e2e/network_quality_metrics_reporter.h +++ b/test/pc/e2e/network_quality_metrics_reporter.h @@ -56,7 +56,7 @@ class NetworkQualityMetricsReporter void ReportPCStats(const std::string& pc_label, const PCStats& stats); void ReportResult(const std::string& metric_name, const std::string& network_label, - const double value, + double value, const std::string& unit) const; std::string GetTestCaseName(const std::string& network_label) const; diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter.h b/test/pc/e2e/stats_based_network_quality_metrics_reporter.h index 709ba7c0e9..50096165fa 100644 --- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.h +++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.h @@ -103,7 +103,7 @@ class StatsBasedNetworkQualityMetricsReporter const Timestamp& end_time); void ReportResult(const std::string& metric_name, const std::string& network_label, - const double value, + double value, const std::string& unit) const; void ReportResult(const std::string& metric_name, const std::string& network_label, diff --git a/test/testsupport/perf_result_reporter.h b/test/testsupport/perf_result_reporter.h index 83e660cd74..a1572341c7 100644 --- a/test/testsupport/perf_result_reporter.h +++ b/test/testsupport/perf_result_reporter.h @@ -81,8 +81,8 @@ class PerfResultReporter { // `mean_and_error` should be a comma-separated string of mean then // error/stddev, e.g. "2.4,0.5". void AddResultMeanAndError(absl::string_view metric_suffix, - const double mean, - const double error); + double mean, + double error); // Returns the metric info if it has been registered. absl::optional GetMetricInfo( diff --git a/test/testsupport/perf_test.h b/test/testsupport/perf_test.h index c5fd34b93f..732fff7d14 100644 --- a/test/testsupport/perf_test.h +++ b/test/testsupport/perf_test.h @@ -49,7 +49,7 @@ enum class ImproveDirection { void PrintResult(absl::string_view measurement, absl::string_view modifier, absl::string_view user_story, - const double value, + double value, absl::string_view units, bool important, ImproveDirection improve_direction = ImproveDirection::kNone); @@ -62,8 +62,8 @@ void PrintResultMeanAndError( absl::string_view measurement, absl::string_view modifier, absl::string_view user_story, - const double mean, - const double error, + double mean, + double error, absl::string_view units, bool important, ImproveDirection improve_direction = ImproveDirection::kNone); @@ -88,7 +88,7 @@ void PrintResult(absl::string_view measurement, absl::string_view user_story, const SamplesStatsCounter& counter, absl::string_view units, - const bool important, + bool important, ImproveDirection improve_direction = ImproveDirection::kNone); // Returns a string-encoded proto as described in diff --git a/test/testsupport/perf_test_result_writer.h b/test/testsupport/perf_test_result_writer.h index e7342c137f..1b93bc9583 100644 --- a/test/testsupport/perf_test_result_writer.h +++ b/test/testsupport/perf_test_result_writer.h @@ -29,24 +29,24 @@ class PerfTestResultWriter { virtual void ClearResults() = 0; virtual void LogResult(absl::string_view graph_name, absl::string_view trace_name, - const double value, + double value, absl::string_view units, - const bool important, + bool important, webrtc::test::ImproveDirection improve_direction) = 0; virtual void LogResultMeanAndError( absl::string_view graph_name, absl::string_view trace_name, - const double mean, - const double error, + double mean, + double error, absl::string_view units, - const bool important, + bool important, webrtc::test::ImproveDirection improve_direction) = 0; virtual void LogResultList( absl::string_view graph_name, absl::string_view trace_name, - const rtc::ArrayView values, + rtc::ArrayView values, absl::string_view units, - const bool important, + bool important, webrtc::test::ImproveDirection improve_direction) = 0; virtual std::string Serialize() const = 0; diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h index a29f186af2..74f3a8631f 100644 --- a/video/video_send_stream_impl.h +++ b/video/video_send_stream_impl.h @@ -77,7 +77,7 @@ class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver, ~VideoSendStreamImpl() override; void DeliverRtcp(const uint8_t* packet, size_t length); - void UpdateActiveSimulcastLayers(const std::vector active_layers); + void UpdateActiveSimulcastLayers(std::vector active_layers); void Start(); void Stop();