Remove top-level const from parameters in function declarations.
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}
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 cc4ed55..d112bfe 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 @@
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_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 @@
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 @@
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 ebb74d6..f106746 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_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 @@
*/
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 b4251ce..ae11394 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 @@
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 @@
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 6b99914..f741e6f 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 @@
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 @@
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 @@
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 @@
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 221e65f..5f4550a 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 @@
* 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 @@
*/
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 @@
*/
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 003d966..8b518fb 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -190,7 +190,7 @@
// 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 a73be2d..076960a 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -36,7 +36,7 @@
// 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 c6e65a0..056dbf5 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 @@
// 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 c96a4d6..fbc42fb 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -28,11 +28,11 @@
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 a3d1520..a9f574b 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -25,11 +25,11 @@
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 @@
~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 @@
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 3ee4dbf..4c50a4b 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -31,11 +31,11 @@
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 7d0cdb9..b7eb9e5 100644
--- a/modules/audio_coding/test/TwoWayCommunication.h
+++ b/modules/audio_coding/test/TwoWayCommunication.h
@@ -31,9 +31,9 @@
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;
diff --git a/modules/audio_device/include/audio_device_data_observer.h b/modules/audio_device/include/audio_device_data_observer.h
index b59cafc..0e93202 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 @@
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 @@
// 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);
diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h
index 01129a4..7b2c784 100644
--- a/modules/audio_device/include/audio_device_defines.h
+++ b/modules/audio_device/include/audio_device_defines.h
@@ -34,21 +34,21 @@
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 8f71a2d..bcba338 100644
--- a/modules/audio_device/include/mock_audio_transport.h
+++ b/modules/audio_device/include/mock_audio_transport.h
@@ -25,23 +25,23 @@
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 1f4a231..23e21d3 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 @@
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 5ed3d47..fe0d3be 100644
--- a/modules/audio_device/mac/audio_device_mac.h
+++ b/modules/audio_device/mac/audio_device_mac.h
@@ -169,16 +169,16 @@
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 17e34ff..0ccab48 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 @@
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 fa7b198..754b4dd 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 @@
// 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 3bfff96..32b564f 100644
--- a/modules/audio_processing/aec3/aec3_common.h
+++ b/modules/audio_processing/aec3/aec3_common.h
@@ -85,10 +85,10 @@
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 2b720a4..7a08217 100644
--- a/modules/audio_processing/aec3/fullband_erle_estimator.h
+++ b/modules/audio_processing/aec3/fullband_erle_estimator.h
@@ -67,7 +67,7 @@
// 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 d6d0d8d..2ee1129 100644
--- a/modules/audio_processing/aecm/aecm_core.h
+++ b/modules/audio_processing/aecm/aecm_core.h
@@ -248,7 +248,7 @@
//
void WebRtcAecm_BufferFarFrame(AecmCore* const aecm,
const int16_t* const farend,
- const int farLen);
+ int farLen);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_FetchFarFrame()
@@ -263,8 +263,8 @@
//
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 @@
//
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 4017037..da3c0cf 100644
--- a/modules/audio_processing/test/fake_recording_device.h
+++ b/modules/audio_processing/test/fake_recording_device.h
@@ -48,8 +48,8 @@
~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 3eee781..16fa657 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 @@
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);
diff --git a/modules/desktop_capture/win/screen_capture_utils.h b/modules/desktop_capture/win/screen_capture_utils.h
index ac60c5a..bcb183b 100644
--- a/modules/desktop_capture/win/screen_capture_utils.h
+++ b/modules/desktop_capture/win/screen_capture_utils.h
@@ -39,24 +39,23 @@
// 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 @@
// 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 8701dc9..b055da7 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 @@
};
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 4b47389..5462650 100644
--- a/modules/video_capture/device_info_impl.h
+++ b/modules/video_capture/device_info_impl.h
@@ -29,7 +29,7 @@
~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 0f60092..3bbe217 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -44,7 +44,7 @@
// 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 93b2a59..b5e9cc6 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 @@
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 @@
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 a3d3fc6..cdfdc60 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 @@
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,
diff --git a/modules/video_processing/util/skin_detection.h b/modules/video_processing/util/skin_detection.h
index 7be791f..7f2e17a 100644
--- a/modules/video_processing/util/skin_detection.h
+++ b/modules/video_processing/util/skin_detection.h
@@ -19,11 +19,11 @@
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