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Remove top-level const from parameters in function declarations.

Browse Source 
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.

Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
This commit is contained in:
Ali Tofigh
2022-01-25 13:27:19 +01:00
committed by WebRTC LUCI CQ
parent c98687a2ef
commit 62238097c9
56 changed files with 235 additions and 238 deletions
Download Patch File Download Diff File Expand all files Collapse all files

24
audio/audio_transport_impl.h
View File

@ -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,

2
call/rtp_payload_params.h
View File

@ -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);

4
call/rtp_video_sender.h
View File

@ -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<bool> active_modules)
void SetActiveModules(std::vector<bool> 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<bool> active_modules)
void SetActiveModulesLocked(std::vector<bool> active_modules)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void ConfigureProtection();

2
call/rtp_video_sender_interface.h
View File

@ -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<bool> active_modules) = 0;
virtual void SetActiveModules(std::vector<bool> active_modules) = 0;
virtual bool IsActive() = 0;
virtual void OnNetworkAvailability(bool network_available) = 0;

4
examples/androidvoip/jni/android_voip_client.h
View File

@ -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<std::string>& 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

6
examples/unityplugin/simple_peer_connection.h
View File

@ -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();

8
examples/unityplugin/unity_plugin_apis.h
View File

@ -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.

2
logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
View File

@ -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();

2
media/base/fake_rtp.h
View File

@ -295,7 +295,7 @@ void CompareHeaderExtensions(const char* packet1,
size_t packet1_size,
const char* packet2,
size_t packet2_size,
const std::vector<int> encrypted_headers,
std::vector<int> encrypted_headers,
bool expect_equal);
#endif // MEDIA_BASE_FAKE_RTP_H_

10
modules/audio_coding/codecs/isac/fix/source/arith_routins.h
View File

@ -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_ */

24
modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
View File

@ -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(...)
@ -101,9 +101,9 @@ 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 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
@ -111,8 +111,8 @@ uint16_t WebRtcIsacfix_GetMinBytes(
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 FrameSamples, /* samples per frame */
int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */
void WebRtcIsacfix_InitRateModel(RateModel* State);

80
modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
View File

@ -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_

8
modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
View File

@ -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

20
modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
View File

@ -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. */
@ -132,9 +132,9 @@ 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 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*/);
@ -144,8 +144,8 @@ 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 FrameSamples, /* samples per frame */
double BottleNeck); /* bottle neck rate; excl headers (bps) */
void WebRtcIsac_InitRateModel(RateModel* State);

2
modules/audio_coding/include/audio_coding_module.h
View File

@ -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;
///////////////////////////////////////////////////////////////////////////

2
modules/audio_coding/neteq/tools/audio_loop.h
View File

@ -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);

2
modules/audio_coding/neteq/tools/input_audio_file.h
View File

@ -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();

8
modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -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:

24
modules/audio_coding/test/RTPFile.h
View File

@ -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,

8
modules/audio_coding/test/TestStereo.h
View File

@ -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();

4
modules/audio_coding/test/TwoWayCommunication.h
View File

@ -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<AudioCodingModule> _acmA;
std::unique_ptr<AudioCodingModule> _acmB;

18
modules/audio_device/include/audio_device_data_observer.h
View File

@ -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<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
// Creates an ADM instance with AudioDeviceDataObserver registered.
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
const AudioDeviceModule::AudioLayer audio_layer,
AudioDeviceModule::AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory,
std::unique_ptr<AudioDeviceDataObserver> observer);
// Creates an ADM instance with AudioDeviceDataObserver registered.
ABSL_DEPRECATED("")
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
const AudioDeviceModule::AudioLayer audio_layer,
AudioDeviceModule::AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory,
AudioDeviceDataObserver* observer);

24
modules/audio_device/include/audio_device_defines.h
View File

@ -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,

24
modules/audio_device/include/mock_audio_transport.h
View File

@ -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,

8
modules/audio_device/linux/audio_device_alsa_linux.h
View File

@ -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;

10
modules/audio_device/mac/audio_device_mac.h
View File

@ -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,

2
modules/audio_device/mac/audio_mixer_manager_mac.h
View File

@ -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);

2
modules/audio_device/win/core_audio_utility_win.h
View File

@ -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.

4
modules/audio_processing/aec3/aec3_common.h
View File

@ -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");

2
modules/audio_processing/aec3/fullband_erle_estimator.h
View File

@ -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.

10
modules/audio_processing/aecm/aecm_core.h
View File

@ -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[];

4
modules/audio_processing/test/fake_recording_device.h
View File

@ -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

2
modules/congestion_controller/goog_cc/test/goog_cc_printer.h
View File

@ -61,7 +61,7 @@ class GoogCcDebugFactory : public GoogCcNetworkControllerFactory {
std::unique_ptr<NetworkControllerInterface> Create(
NetworkControllerConfig config) override;
void PrintState(const Timestamp at_time);
void PrintState(Timestamp at_time);
void AttachWriter(std::unique_ptr<RtcEventLogOutput> log_writer);

11
modules/desktop_capture/win/screen_capture_utils.h
View File

@ -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

14
modules/desktop_capture/win/test_support/test_window.h
View File

@ -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);

2
modules/video_capture/device_info_impl.h
View File

@ -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,

2
modules/video_capture/video_capture.h
View File

@ -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

9
modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
View File

@ -69,11 +69,10 @@ class LibvpxVp9Encoder : public VP9Encoder {
absl::optional<int>* 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

6
modules/video_coding/utility/simulcast_test_fixture_impl.h
View File

@ -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<bool> active_streams);
void UpdateActiveStreams(const std::vector<bool> active_streams);
void RunActiveStreamsTest(std::vector<bool> active_streams);
void UpdateActiveStreams(std::vector<bool> active_streams);
void ExpectStreams(VideoFrameType frame_type,
const std::vector<bool> expected_streams_active);
std::vector<bool> expected_streams_active);
void ExpectStreams(VideoFrameType frame_type, int expected_video_streams);
void VerifyTemporalIdxAndSyncForAllSpatialLayers(
TestEncodedImageCallback* encoder_callback,

10
modules/video_processing/util/skin_detection.h
View File

@ -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

3
net/dcsctp/rx/traditional_reassembly_streams.h
View File

@ -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_;
};

4
net/dcsctp/testing/data_generator.h
View File

@ -38,14 +38,14 @@ class DataGenerator {
// "is_end" flag.
Data Ordered(std::vector<uint8_t> 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<uint8_t> 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); }

2
p2p/stunprober/stun_prober.h
View File

@ -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.

2
pc/jsep_transport_controller.h
View File

@ -135,7 +135,7 @@ class JsepTransportController : public sigslot::has_slots<> {
// Factory for SCTP transports.
SctpTransportFactoryInterface* sctp_factory = nullptr;
std::function<void(const rtc::SSLHandshakeError)> on_dtls_handshake_error_;
std::function<void(rtc::SSLHandshakeError)> on_dtls_handshake_error_;
};
// The ICE related events are fired on the `network_thread`.

2
pc/rtp_transmission_manager.h
View File

@ -184,7 +184,7 @@ class RtpTransmissionManager : public RtpSenderBase::SetStreamsObserver {
cricket::MediaType media_type);
const RtpSenderInfo* FindSenderInfo(const std::vector<RtpSenderInfo>& 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<RtpSenderProxyWithInternal<RtpSenderInternal>>

2
rtc_base/experiments/struct_parameters_parser.h
View File

@ -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);
};

2
rtc_base/string_encode.h
View File

@ -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);

2
rtc_base/string_utils.h
View File

@ -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

2
sdk/objc/components/video_codec/helpers.h
View File

@ -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);

2
test/direct_transport.h
View File

@ -32,7 +32,7 @@ class Demuxer {
explicit Demuxer(const std::map<uint8_t, MediaType>& 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<uint8_t, MediaType> payload_type_map_;
RTC_DISALLOW_COPY_AND_ASSIGN(Demuxer);
};

2
test/pc/e2e/network_quality_metrics_reporter.h
View File

@ -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;

2
test/pc/e2e/stats_based_network_quality_metrics_reporter.h
View File

@ -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,

4
test/testsupport/perf_result_reporter.h
View File

@ -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<MetricInfo> GetMetricInfo(

8
test/testsupport/perf_test.h
View File

@ -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

14
test/testsupport/perf_test_result_writer.h
View File

@ -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<const double> values,
rtc::ArrayView<const double> values,
absl::string_view units,
const bool important,
bool important,
webrtc::test::ImproveDirection improve_direction) = 0;
virtual std::string Serialize() const = 0;

2
video/video_send_stream_impl.h
View File

@ -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<bool> active_layers);
void UpdateActiveSimulcastLayers(std::vector<bool> active_layers);
void Start();
void Stop();