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;