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Upconvert various types to int.

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Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.

Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."

This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t).  Other locations will be converted to size_t in a separate change.

BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54629004

Cr-Commit-Position: refs/heads/master@{#9405}
This commit is contained in:
Peter Kasting
2015-06-09 17:19:57 -07:00
parent 54b0ca553f
commit 83ad33a8ae
61 changed files with 314 additions and 314 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
View File

@ -68,8 +68,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
* -1 - Error
*/
int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
int16_t quality);
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality);
int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
/****************************************************************************
@ -103,9 +103,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID);
int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID);
/****************************************************************************
* WebRtcCng_UpdateSid(...)

18
webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
View File

@ -36,7 +36,7 @@ typedef struct WebRtcCngDecoder_ {
typedef struct WebRtcCngEncoder_ {
int16_t enc_nrOfCoefs;
uint16_t enc_sampfreq;
int enc_sampfreq;
int16_t enc_interval;
int16_t enc_msSinceSID;
int32_t enc_Energy;
@ -142,8 +142,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
int16_t quality) {
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality) {
int i;
WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
memset(inst, 0, sizeof(WebRtcCngEncoder));
@ -227,9 +227,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst) {
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID) {
int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID) {
WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@ -388,10 +388,12 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
inst->enc_msSinceSID = 0;
*bytesOut = inst->enc_nrOfCoefs + 1;
inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
inst->enc_msSinceSID +=
(int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
return inst->enc_nrOfCoefs + 1;
} else {
inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
inst->enc_msSinceSID +=
(int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
*bytesOut = 0;
return 0;
}

4
webrtc/modules/audio_coding/codecs/g722/g722_interface.c
View File

@ -39,7 +39,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
}
}
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
// Free encoder memory
return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
@ -79,7 +79,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
}
}
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
// Free encoder memory
return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);

4
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@ -73,7 +73,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
@ -142,7 +142,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
* -1 - Error
*/
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
/****************************************************************************

2
webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
View File

@ -31,7 +31,7 @@ void WebRtcIlbcfix_AugmentedCbCorr(
int16_t low, /* (i) Lag to start from (typically
20) */
int16_t high, /* (i) Lag to end at (typically 39) */
int16_t scale) /* (i) Scale factor to use for
int scale) /* (i) Scale factor to use for
the crossDot */
{
int lagcount;

3
webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
View File

@ -36,7 +36,6 @@ void WebRtcIlbcfix_AugmentedCbCorr(
int16_t low, /* (i) Lag to start from (typically
20) */
int16_t high, /* (i) Lag to end at (typically 39 */
int16_t scale); /* (i) Scale factor to use for
the crossDot */
int scale); /* (i) Scale factor to use for the crossDot */
#endif

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
View File

@ -34,7 +34,7 @@ void WebRtcIlbcfix_CbMemEnergy(
int16_t lTarget, /* (i) Length of the target vector */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts, /* (o) Shift value of the energy */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size /* (i) Index to where the energy values should be stored */
) {
int16_t *ppi, *ppo, *pp;

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
View File

@ -27,7 +27,7 @@ void WebRtcIlbcfix_CbMemEnergy(
int16_t lTarget, /* (i) Length of the target vector */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts, /* (o) Shift value of the energy */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size /* (i) Index to where the energy values should be stored */
);

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
View File

@ -22,7 +22,7 @@
void WebRtcIlbcfix_CbMemEnergyAugmentation(
int16_t *interpSamples, /* (i) The interpolated samples */
int16_t *CBmem, /* (i) The CB memory */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size, /* (i) Index to where the energy values should be stored */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts /* (o) Shift value of the energy */

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
View File

@ -22,7 +22,7 @@
void WebRtcIlbcfix_CbMemEnergyAugmentation(
int16_t *interpSamples, /* (i) The interpolated samples */
int16_t *CBmem, /* (i) The CB memory */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size, /* (i) Index to where the energy values should be stored */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts /* (o) Shift value of the energy */

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
View File

@ -28,7 +28,7 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
int16_t *ppo, /* (i) input pointer 2 */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts, /* (o) Shift value of the energy */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size /* (i) Index to where the energy values should be stored */
)
{

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
View File

@ -26,7 +26,7 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
int16_t *ppo, /* (i) input pointer 2 */
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts, /* (o) Shift value of the energy */
int16_t scale, /* (i) The scaling of all energy values */
int scale, /* (i) The scaling of all energy values */
int16_t base_size /* (i) Index to where the energy values should be stored */
);

4
webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
View File

@ -46,7 +46,9 @@ void WebRtcIlbcfix_CbSearch(
int16_t block /* (i) the subblock number */
) {
int16_t i, j, stage, range;
int16_t *pp, scale, tmp;
int16_t *pp;
int16_t tmp;
int scale;
int16_t bits, temp1, temp2;
int16_t base_size;
int32_t codedEner, targetEner;

8
webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
View File

@ -121,8 +121,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
shifts = WEBRTC_SPL_MAX(0, shifts);
/* compute cross correlation */
WebRtcSpl_CrossCorrelation(corr32, target, regressor,
ENH_BLOCKL_HALF, 50, (int16_t)shifts, -1);
WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
shifts, -1);
/* Find 3 highest correlations that should be compared for the
highest (corr*corr)/ener */
@ -207,8 +207,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
shifts=0;
/* compute cross correlation */
WebRtcSpl_CrossCorrelation(corr32, target, regressor,
plc_blockl, 3, (int16_t)shifts, 1);
WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
1);
/* find lag */
lag=WebRtcSpl_MaxIndexW32(corr32, 3);

38
webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
View File

@ -88,10 +88,10 @@ int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
}
}
int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
const int16_t* speechIn,
int16_t len,
uint8_t* encoded) {
int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
const int16_t* speechIn,
int16_t len,
uint8_t* encoded) {
int16_t pos = 0;
int16_t encpos = 0;
@ -141,11 +141,11 @@ int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
}
int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
{
int i=0;
/* Allow for automatic switching between the frame sizes
@ -194,11 +194,11 @@ int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
}
int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
{
int i=0;
if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
@ -222,11 +222,11 @@ int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
}
int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType)
{
int i=0;
if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||

2
webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
View File

@ -23,7 +23,7 @@
* Initiation of decoder instance.
*---------------------------------------------------------------*/
int16_t WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
int16_t mode, /* (i) frame size mode */
int use_enhancer) { /* (i) 1: use enhancer, 0: no enhancer */

2
webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
View File

@ -25,7 +25,7 @@
* Initiation of decoder instance.
*---------------------------------------------------------------*/
int16_t WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
int16_t mode, /* (i) frame size mode */
int use_enhancer /* (i) 1 to use enhancer

2
webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
View File

@ -23,7 +23,7 @@
* Initiation of encoder instance.
*---------------------------------------------------------------*/
int16_t WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
int16_t mode) { /* (i) frame size mode */
iLBCenc_inst->mode = mode;

2
webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
View File

@ -25,7 +25,7 @@
* Initiation of encoder instance.
*---------------------------------------------------------------*/
int16_t WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
int16_t mode /* (i) frame size mode */
);

30
webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
View File

@ -135,10 +135,10 @@ extern "C" {
* -1 - Error
*/
int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
const int16_t *speechIn,
int16_t len,
uint8_t* encoded);
int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
const int16_t *speechIn,
int16_t len,
uint8_t* encoded);
/****************************************************************************
* WebRtcIlbcfix_DecoderInit(...)
@ -180,21 +180,21 @@ extern "C" {
* -1 - Error
*/
int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
/****************************************************************************
* WebRtcIlbcfix_DecodePlc(...)

5
webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
View File

@ -29,7 +29,8 @@ void WebRtcIlbcfix_MyCorr(
const int16_t *seq2, /* (i) second sequence */
int16_t dim2 /* (i) dimension seq2 */
){
int16_t max, scale, loops;
int16_t max, loops;
int scale;
/* Calculate correlation between the two sequences. Scale the
result of the multiplcication to maximum 26 bits in order
@ -37,7 +38,7 @@ void WebRtcIlbcfix_MyCorr(
max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
scale=WebRtcSpl_GetSizeInBits(max);
scale = (int16_t)(2 * scale - 26);
scale = 2 * scale - 26;
if (scale<0) {
scale=0;
}

6
webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
View File

@ -41,7 +41,8 @@ int main(int argc, char* argv[])
{
FILE *ifileid,*efileid,*ofileid, *chfileid;
short encoded_data[55], data[240], speechType;
short len, mode, pli;
int len;
short mode, pli;
int blockcount = 0;
IlbcEncoderInstance *Enc_Inst;
@ -173,7 +174,8 @@ int main(int argc, char* argv[])
/* decoding */
fprintf(stderr, "--- Decoding block %i --- ",blockcount);
if (pli==1) {
len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, &speechType);
len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, (int16_t)len, data,
&speechType);
} else {
len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
}

2
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View File

@ -184,7 +184,7 @@ int AudioEncoderDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
decoder_sample_rate_hz_ = sample_rate_hz;
}
int16_t temp_type = 1; // Default is speech.
int16_t ret =
int ret =
T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);

20
webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
View File

@ -25,12 +25,12 @@ struct IsacFix {
static const uint16_t kFixSampleRate = 16000;
static inline int16_t Control(instance_type* inst,
int32_t rate,
int16_t framesize) {
int framesize) {
return WebRtcIsacfix_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
int16_t frame_size_ms,
int frame_size_ms,
int16_t enforce_frame_size) {
return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
enforce_frame_size);
@ -38,11 +38,11 @@ struct IsacFix {
static inline int16_t Create(instance_type** inst) {
return WebRtcIsacfix_Create(inst);
}
static inline int16_t DecodeInternal(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
static inline int DecodeInternal(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
}
static inline int16_t DecodePlc(instance_type* inst,
@ -53,9 +53,9 @@ struct IsacFix {
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsacfix_DecoderInit(inst);
}
static inline int16_t Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
static inline int Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
return WebRtcIsacfix_Encode(inst, speech_in, encoded);
}
static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {

32
webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
View File

@ -128,9 +128,9 @@ extern "C" {
* -1 - Error
*/
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
uint8_t* encoded);
int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
uint8_t* encoded);
@ -251,11 +251,11 @@ extern "C" {
* -1 - Error
*/
int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);
int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);
/****************************************************************************
@ -280,11 +280,11 @@ extern "C" {
*/
#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);
int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);
#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
@ -378,8 +378,8 @@ extern "C" {
*/
int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rate,
int16_t framesize);
int16_t rate,
int framesize);
@ -407,7 +407,7 @@ extern "C" {
int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rateBPS,
int16_t frameSizeMs,
int frameSizeMs,
int16_t enforceFrameSize);

8
webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
View File

@ -226,10 +226,10 @@ int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
* Return value : number of bytes in the stream so far
* -1 if error detected
*/
int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
Bitstr_dec *streamData,
const int32_t *envQ8,
const int16_t lenData)
int WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
Bitstr_dec *streamData,
const int32_t *envQ8,
const int16_t lenData)
{
uint32_t W_lower;
uint32_t W_upper;

2
webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
View File

@ -74,7 +74,7 @@ int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
* Return value : number of bytes in the stream so far
* <0 if error detected
*/
int16_t WebRtcIsacfix_DecLogisticMulti2(
int WebRtcIsacfix_DecLogisticMulti2(
int16_t *data,
Bitstr_dec *streamData,
const int32_t *env,

6
webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
View File

@ -32,9 +32,9 @@ int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr* bwest_str,
uint32_t send_ts,
uint32_t arr_ts);
int16_t WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
IsacFixDecoderInstance* ISACdec_obj,
int16_t* current_framesamples);
int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
IsacFixDecoderInstance* ISACdec_obj,
int16_t* current_framesamples);
int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
IsacFixDecoderInstance* ISACdec_obj,

8
webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
View File

@ -27,14 +27,14 @@
int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
IsacFixDecoderInstance *ISACdec_obj,
int16_t *current_framesamples)
int WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
IsacFixDecoderInstance *ISACdec_obj,
int16_t *current_framesamples)
{
int k;
int err;
int16_t BWno;
int16_t len = 0;
int len = 0;
int16_t model;

10
webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
View File

@ -450,10 +450,10 @@ static void GenerateDitherQ7(int16_t *bufQ7,
* function to decode the complex spectrum from the bitstream
* returns the total number of bytes in the stream
*/
int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
int16_t *frQ7,
int16_t *fiQ7,
int16_t AvgPitchGain_Q12)
int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
int16_t *frQ7,
int16_t *fiQ7,
int16_t AvgPitchGain_Q12)
{
int16_t data[FRAMESAMPLES];
int32_t invARSpec2_Q16[FRAMESAMPLES/4];
@ -461,7 +461,7 @@ int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
int16_t RCQ15[AR_ORDER];
int16_t gainQ10;
int32_t gain2_Q10;
int16_t len;
int len;
int k;
/* create dither signal */

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

@ -22,10 +22,10 @@
#include "structs.h"
/* decode complex spectrum (return number of bytes in stream) */
int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
int16_t *frQ7,
int16_t *fiQ7,
int16_t AvgPitchGain_Q12);
int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
int16_t *frQ7,
int16_t *fiQ7,
int16_t AvgPitchGain_Q12);
/* encode complex spectrum */
int WebRtcIsacfix_EncodeSpec(const int16_t *fr,

48
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@ -399,12 +399,12 @@ static void write_be16(const uint16_t* src, size_t nbytes, uint8_t* dest) {
* : -1 - Error
*/
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
uint8_t* encoded)
int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
uint8_t* encoded)
{
ISACFIX_SubStruct *ISAC_inst;
int16_t stream_len;
int stream_len;
/* typecast pointer to rela structure */
ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@ -421,7 +421,7 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
&ISAC_inst->bwestimator_obj,
ISAC_inst->CodingMode);
if (stream_len<0) {
ISAC_inst->errorcode = - stream_len;
ISAC_inst->errorcode = -(int16_t)stream_len;
return -1;
}
@ -766,17 +766,17 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
*/
int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
{
ISACFIX_SubStruct *ISAC_inst;
/* number of samples (480 or 960), output from decoder */
/* that were actually used in the encoder/decoder (determined on the fly) */
int16_t number_of_samples;
int16_t declen = 0;
int declen = 0;
/* typecast pointer to real structure */
ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@ -809,7 +809,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
if (declen < 0) {
/* Some error inside the decoder */
ISAC_inst->errorcode = -declen;
ISAC_inst->errorcode = -(int16_t)declen;
memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES);
return -1;
}
@ -859,17 +859,17 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
*/
#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
{
ISACFIX_SubStruct *ISAC_inst;
/* twice the number of samples (480 or 960), output from decoder */
/* that were actually used in the encoder/decoder (determined on the fly) */
int16_t number_of_samples;
int16_t declen = 0;
int declen = 0;
int16_t dummy[FRAMESAMPLES/2];
@ -903,7 +903,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
if (declen < 0) {
/* Some error inside the decoder */
ISAC_inst->errorcode = -declen;
ISAC_inst->errorcode = -(int16_t)declen;
memset(decoded, 0, sizeof(int16_t) * FRAMESAMPLES);
return -1;
}
@ -1076,8 +1076,8 @@ int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
*/
int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rate,
int16_t framesize)
int16_t rate,
int framesize)
{
ISACFIX_SubStruct *ISAC_inst;
/* typecast pointer to real structure */
@ -1101,7 +1101,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
if (framesize == 30 || framesize == 60)
ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * framesize;
ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * framesize);
else {
ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
return -1;
@ -1136,7 +1136,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rateBPS,
int16_t frameSizeMs,
int frameSizeMs,
int16_t enforceFrameSize)
{
ISACFIX_SubStruct *ISAC_inst;
@ -1170,7 +1170,7 @@ int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
/* Set initial framesize. If enforceFrameSize is set the frame size will not change */
if ((frameSizeMs == 30) || (frameSizeMs == 60)) {
ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * frameSizeMs;
ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * frameSizeMs);
} else {
ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
return -1;

7
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
View File

@ -101,14 +101,15 @@ int main(int argc, char* argv[])
int i, errtype, h = 0, k, packetLossPercent = 0;
int16_t CodingMode;
int16_t bottleneck;
int16_t framesize = 30; /* ms */
int framesize = 30; /* ms */
int cur_framesmpls, err = 0, lostPackets = 0;
/* Runtime statistics */
double starttime, runtime, length_file;
int16_t stream_len = 0;
int16_t framecnt, declen = 0;
int16_t framecnt;
int declen = 0;
int16_t shortdata[FRAMESAMPLES_10ms];
int16_t decoded[MAX_FRAMESAMPLES];
uint16_t streamdata[500];
@ -766,7 +767,7 @@ int main(int argc, char* argv[])
#else
declen = -1;
#endif
prevFrameSize = declen/240;
prevFrameSize = static_cast<int16_t>(declen / 240);
}
}

6
webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
View File

@ -88,8 +88,8 @@ int main(int argc, char* argv[]) {
int16_t CodingMode;
int16_t bottleneck;
int16_t framesize = 30; /* ms */
// int16_t framesize = 60; /* To invoke cisco complexity case at frame 2252 */
int framesize = 30; /* ms */
// int framesize = 60; /* To invoke cisco complexity case at frame 2252 */
int cur_framesmpls, err;
@ -99,7 +99,7 @@ int main(int argc, char* argv[]) {
double length_file;
int16_t stream_len = 0;
int16_t declen;
int declen;
int16_t shortdata[FRAMESAMPLES_10ms];
int16_t decoded[MAX_FRAMESAMPLES];

20
webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
View File

@ -24,12 +24,12 @@ struct IsacFloat {
static const bool has_swb = true;
static inline int16_t Control(instance_type* inst,
int32_t rate,
int16_t framesize) {
int framesize) {
return WebRtcIsac_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
int16_t frame_size_ms,
int frame_size_ms,
int16_t enforce_frame_size) {
return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
enforce_frame_size);
@ -37,11 +37,11 @@ struct IsacFloat {
static inline int16_t Create(instance_type** inst) {
return WebRtcIsac_Create(inst);
}
static inline int16_t DecodeInternal(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
static inline int DecodeInternal(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
}
static inline int16_t DecodePlc(instance_type* inst,
@ -53,9 +53,9 @@ struct IsacFloat {
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsac_DecoderInit(inst);
}
static inline int16_t Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
static inline int Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
return WebRtcIsac_Encode(inst, speech_in, encoded);
}
static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {

10
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
View File

@ -144,7 +144,7 @@ extern "C" {
* : -1 - Error
*/
int16_t WebRtcIsac_Encode(
int WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded);
@ -214,7 +214,7 @@ extern "C" {
* -1 - Error.
*/
int16_t WebRtcIsac_Decode(
int WebRtcIsac_Decode(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
@ -269,7 +269,7 @@ extern "C" {
int16_t WebRtcIsac_Control(
ISACStruct* ISAC_main_inst,
int32_t rate,
int16_t framesize);
int framesize);
/******************************************************************************
@ -300,7 +300,7 @@ extern "C" {
int16_t WebRtcIsac_ControlBwe(
ISACStruct* ISAC_main_inst,
int32_t rateBPS,
int16_t frameSizeMs,
int frameSizeMs,
int16_t enforceFrameSize);
@ -701,7 +701,7 @@ extern "C" {
* Return value : >0 - number of samples in decoded vector
* -1 - Error
*/
int16_t WebRtcIsac_DecodeRcu(
int WebRtcIsac_DecodeRcu(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t len,

6
webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
View File

@ -80,9 +80,9 @@ static const uint32_t kCrcTable[256] = {
* -1 - Error
*/
int16_t WebRtcIsac_GetCrc(const int16_t* bitstream,
int16_t len_bitstream_in_bytes,
uint32_t* crc)
int WebRtcIsac_GetCrc(const int16_t* bitstream,
int len_bitstream_in_bytes,
uint32_t* crc)
{
uint8_t* bitstream_ptr_uw8;
uint32_t crc_state;

6
webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
View File

@ -36,10 +36,10 @@
* -1 - Error
*/
int16_t WebRtcIsac_GetCrc(
int WebRtcIsac_GetCrc(
const int16_t* encoded,
int16_t no_of_word8s,
uint32_t* crc);
int no_of_word8s,
uint32_t* crc);

62
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -494,15 +494,15 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
* samples.
* : -1 - Error
*/
int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded) {
int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded) {
float inFrame[FRAMESAMPLES_10ms];
int16_t speechInLB[FRAMESAMPLES_10ms];
int16_t speechInUB[FRAMESAMPLES_10ms];
int16_t streamLenLB = 0;
int16_t streamLenUB = 0;
int16_t streamLen = 0;
int streamLenLB = 0;
int streamLenUB = 0;
int streamLen = 0;
int16_t k = 0;
int garbageLen = 0;
int32_t bottleneck = 0;
@ -601,8 +601,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
/* Tell to upper-band the number of bytes used so far.
* This is for payload limitation. */
instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
LEN_CHECK_SUM_WORD8;
instUB->ISACencUB_obj.numBytesUsed =
(int16_t)(streamLenLB + 1 + LEN_CHECK_SUM_WORD8);
/* Encode upper-band. */
switch (instISAC->bandwidthKHz) {
case isac12kHz: {
@ -1045,12 +1045,12 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
return 0;
}
static int16_t Decode(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType,
int16_t isRCUPayload) {
static int Decode(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType,
int16_t isRCUPayload) {
/* Number of samples (480 or 960), output from decoder
that were actually used in the encoder/decoder
(determined on the fly). */
@ -1060,8 +1060,8 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
float outFrame[MAX_FRAMESAMPLES];
int16_t outFrameLB[MAX_FRAMESAMPLES];
int16_t outFrameUB[MAX_FRAMESAMPLES];
int16_t numDecodedBytesLB;
int16_t numDecodedBytesUB;
int numDecodedBytesLB;
int numDecodedBytesUB;
int16_t lenEncodedLBBytes;
int16_t validChecksum = 1;
int16_t k;
@ -1350,11 +1350,11 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
* -1 - Error
*/
int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {
int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {
int16_t isRCUPayload = 0;
return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
speechType, isRCUPayload);
@ -1382,11 +1382,11 @@ int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {
int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {
int16_t isRCUPayload = 1;
return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
speechType, isRCUPayload);
@ -1485,7 +1485,7 @@ static int16_t ControlUb(ISACUBStruct* instISAC, double rate) {
int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
int32_t bottleneckBPS,
int16_t frameSize) {
int frameSize) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
int16_t status;
double rateLB;
@ -1526,7 +1526,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
return -1;
}
status = ControlLb(&instISAC->instLB, rateLB, frameSize);
status = ControlLb(&instISAC->instLB, rateLB, (int16_t)frameSize);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
@ -1594,7 +1594,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
*/
int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
int32_t bottleneckBPS,
int16_t frameSizeMs,
int frameSizeMs,
int16_t enforceFrameSize) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum ISACBandwidth bandwidth;
@ -1641,8 +1641,8 @@ int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
* will not change */
if (frameSizeMs != 0) {
if ((frameSizeMs == 30) || (frameSizeMs == 60)) {
instISAC->instLB.ISACencLB_obj.new_framelength = (FS / 1000) *
frameSizeMs;
instISAC->instLB.ISACencLB_obj.new_framelength =
(int16_t)((FS / 1000) * frameSizeMs);
} else {
instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
return -1;

2
webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
View File

@ -79,7 +79,7 @@ TEST_F(IsacTest, IsacUpdateBWE) {
WebRtcIsac_EncoderInit(isac_codec_, 0);
WebRtcIsac_DecoderInit(isac_codec_);
int16_t encoded_bytes;
int encoded_bytes;
// Test with call with a small packet (sync packet).
EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1,

5
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -47,14 +47,15 @@ int main(int argc, char* argv[]) {
int i, errtype, VADusage = 0, packetLossPercent = 0;
int16_t CodingMode;
int32_t bottleneck = 0;
int16_t framesize = 30; /* ms */
int framesize = 30; /* ms */
int cur_framesmpls, err;
/* Runtime statistics */
double starttime, runtime, length_file;
int16_t stream_len = 0;
int16_t declen = 0, lostFrame = 0, declenTC = 0;
int declen = 0, declenTC = 0;
int16_t lostFrame = 0;
int16_t shortdata[SWBFRAMESAMPLES_10ms];
int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];

2
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@ -191,7 +191,7 @@ int main(int argc, char* argv[])
short streamLen;
short numSamplesRead;
short lenDecodedAudio;
int lenDecodedAudio;
short senderIdx;
short receiverIdx;

6
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -62,7 +62,7 @@ int main(int argc, char* argv[]) {
unsigned long totalsmpls = 0;
int32_t bottleneck = 39;
int16_t frameSize = 30; /* ms */
int frameSize = 30; /* ms */
int16_t codingMode = 1;
int16_t shortdata[FRAMESAMPLES_SWB_10ms];
int16_t decoded[MAX_FRAMESAMPLES_SWB];
@ -73,9 +73,9 @@ int main(int argc, char* argv[]) {
ISACStruct* ISAC_main_inst;
int16_t stream_len = 0;
int16_t declen = 0;
int declen = 0;
int16_t err;
int16_t cur_framesmpls;
int cur_framesmpls;
int endfile;
#ifdef WIN32
double length_file;

2
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -198,7 +198,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
CHECK_EQ(input_buffer_.size(),
static_cast<size_t>(num_10ms_frames_per_packet_) *
samples_per_10ms_frame_);
int16_t status = WebRtcOpus_Encode(
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
static_cast<int16_t>(num_channels_)),

26
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@ -64,11 +64,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
* Return value : >=0 - Length (in bytes) of coded data
* -1 - Error
*/
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
int WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
/****************************************************************************
* WebRtcOpus_SetBitRate(...)
@ -236,9 +236,9 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
* Return value : >0 - Samples per channel in decoded vector
* -1 - Error
*/
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DecodePlc(...)
@ -254,8 +254,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DecodeFec(...)
@ -275,9 +275,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
* 0 - No FEC data in the packet
* -1 - Error
*/
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DurationEst(...)

10
webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -131,10 +131,10 @@ OpusFecTest::OpusFecTest()
}
void OpusFecTest::EncodeABlock() {
int16_t value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
int value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
EXPECT_GT(value, 0);
encoded_bytes_ = value;
@ -142,7 +142,7 @@ void OpusFecTest::EncodeABlock() {
void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
int16_t audio_type;
int16_t value_1 = 0, value_2 = 0;
int value_1 = 0, value_2 = 0;
if (lost_previous) {
// Decode previous frame.

26
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -78,11 +78,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
}
}
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int res;
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
@ -291,9 +291,9 @@ static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
return res;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
if (encoded_bytes == 0) {
@ -318,8 +318,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
return decoded_samples;
}
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames) {
int16_t audio_type = 0;
int decoded_samples;
int plc_samples;
@ -339,9 +339,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
return decoded_samples;
}
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
int fec_samples;