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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

10
modules/audio_coding/codecs/isac/fix/source/arith_routins.h
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@ -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_ */

28
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(...)
@ -100,19 +100,19 @@ int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str);
*/
uint16_t WebRtcIsacfix_GetMinBytes(
RateModel* State,
int16_t StreamSize, /* bytes in bitstream */
const int16_t FrameLen, /* ms per frame */
const int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */
const int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */
int16_t StreamSize, /* bytes in bitstream */
int16_t FrameLen, /* ms per frame */
int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */
int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */
/*
* update long-term average bitrate and amount of data in buffer
*/
void WebRtcIsacfix_UpdateRateModel(
RateModel* State,
int16_t StreamSize, /* bytes in bitstream */
const int16_t FrameSamples, /* samples per frame */
const int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */
int16_t StreamSize, /* bytes in bitstream */
int16_t FrameSamples, /* samples per frame */
int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */
void WebRtcIsacfix_InitRateModel(RateModel* State);

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

24
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. */
@ -131,10 +131,10 @@ int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str);
*/
int WebRtcIsac_GetMinBytes(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameLen, /* ms per frame */
const double BottleNeck, /* bottle neck rate; excl headers (bps) */
const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
int StreamSize, /* bytes in bitstream */
int FrameLen, /* ms per frame */
double BottleNeck, /* bottle neck rate; excl headers (bps) */
double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
enum ISACBandwidth bandwidth
/*,int16_t frequentLargePackets*/);
@ -143,9 +143,9 @@ int WebRtcIsac_GetMinBytes(
*/
void WebRtcIsac_UpdateRateModel(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameSamples, /* samples per frame */
const double BottleNeck); /* bottle neck rate; excl headers (bps) */
int StreamSize, /* bytes in bitstream */
int FrameSamples, /* samples per frame */
double BottleNeck); /* bottle neck rate; excl headers (bps) */
void WebRtcIsac_InitRateModel(RateModel* State);

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;