Upconvert various types to int.
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}
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
index b016f40..1ec5d67 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
@@ -68,8 +68,8 @@
* -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 @@
* 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(...)
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 9862f12..32e2859 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -36,7 +36,7 @@
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 @@
* 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 @@
* 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 @@
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;
}
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index d06c588..6a669e2 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -39,7 +39,7 @@
}
}
-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_FreeDecoder(G722DecInst *G722dec_inst)
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
// Free encoder memory
return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
index 7fe11a7..a5ecbe7 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -73,7 +73,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
@@ -142,7 +142,7 @@
* -1 - Error
*/
-int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
/****************************************************************************
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
index d8f8c93..c24b4a6 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -31,7 +31,7 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
index 533d0a4..a0435c4 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -36,7 +36,6 @@
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
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
index f8a0933..2b7e082 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -34,7 +34,7 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 1b50c0b..68dd7da 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -27,7 +27,7 @@
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 */
);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
index 7e6daf9..39f18c2 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -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 */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
index 6c181bd..e73d414 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -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 */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
index e5d1424..b675441 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -28,7 +28,7 @@
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 */
)
{
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
index c7e1e54..c7bf929 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -26,7 +26,7 @@
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 */
);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
index 4c6196b..2a77f4f 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -46,7 +46,9 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index fde5414..db4b8d4 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -121,8 +121,8 @@
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 @@
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);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index 88ad33b..e41c095 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -88,10 +88,10 @@
}
}
-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_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 @@
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 @@
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)||
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
index d903ac7..0659e50 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -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 */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
index 4871b5c..cdd2192 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -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
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
index 1a2fa08..9c562db 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -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;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
index 2eea27c..7154661 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -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 */
);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
index b7e1735..4934968 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
@@ -135,10 +135,10 @@
* -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 @@
* -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);
- 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);
+ int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+ const uint8_t* encoded,
+ int16_t len,
+ int16_t* decoded,
+ int16_t* speechType);
/****************************************************************************
* WebRtcIlbcfix_DecodePlc(...)
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
index 048745a..0da6d54 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -29,7 +29,8 @@
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 @@
max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
scale=WebRtcSpl_GetSizeInBits(max);
- scale = (int16_t)(2 * scale - 26);
+ scale = 2 * scale - 26;
if (scale<0) {
scale=0;
}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index 370bf9d..1e966a7 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -41,7 +41,8 @@
{
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 @@
/* 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);
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 65c5b90..befb355 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -184,7 +184,7 @@
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);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
index bf9f875..a1eb271 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
@@ -25,12 +25,12 @@
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 @@
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 @@
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) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
index 961fd3f..92dcf51 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -128,9 +128,9 @@
* -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 @@
* -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 @@
*/
#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 @@
*/
int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
- int16_t rate,
- int16_t framesize);
+ int16_t rate,
+ int framesize);
@@ -407,7 +407,7 @@
int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rateBPS,
- int16_t frameSizeMs,
+ int frameSizeMs,
int16_t enforceFrameSize);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index 23048a5..808aeb7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -226,10 +226,10 @@
* 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
index 584bc47..40bbb4c 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
@@ -74,7 +74,7 @@
* 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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
index 8ecbd14..3d003e4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -32,9 +32,9 @@
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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index 5e095da..f0ae07e 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 3965378..aab8f43 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -450,10 +450,10 @@
* 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 RCQ15[AR_ORDER];
int16_t gainQ10;
int32_t gain2_Q10;
- int16_t len;
+ int len;
int k;
/* create dither signal */
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index ec20c71..e4489df 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index f8abc8a..f1e5cd0 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -399,12 +399,12 @@
* : -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 @@
&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_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 @@
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 @@
*/
#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 @@
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_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 @@
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_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 @@
/* 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index ba50b0c..bc1a194 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -101,14 +101,15 @@
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 @@
#else
declen = -1;
#endif
- prevFrameSize = declen/240;
+ prevFrameSize = static_cast<int16_t>(declen / 240);
}
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 9aabd04..218b97b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -88,8 +88,8 @@
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 @@
double length_file;
int16_t stream_len = 0;
- int16_t declen;
+ int declen;
int16_t shortdata[FRAMESAMPLES_10ms];
int16_t decoded[MAX_FRAMESAMPLES];
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
index 5a75807..8c70533 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
@@ -24,12 +24,12 @@
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 @@
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 @@
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) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
index 6d0c32d..1a83d72 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -144,7 +144,7 @@
* : -1 - Error
*/
- int16_t WebRtcIsac_Encode(
+ int WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded);
@@ -214,7 +214,7 @@
* -1 - Error.
*/
- int16_t WebRtcIsac_Decode(
+ int WebRtcIsac_Decode(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
@@ -269,7 +269,7 @@
int16_t WebRtcIsac_Control(
ISACStruct* ISAC_main_inst,
int32_t rate,
- int16_t framesize);
+ int framesize);
/******************************************************************************
@@ -300,7 +300,7 @@
int16_t WebRtcIsac_ControlBwe(
ISACStruct* ISAC_main_inst,
int32_t rateBPS,
- int16_t frameSizeMs,
+ int frameSizeMs,
int16_t enforceFrameSize);
@@ -701,7 +701,7 @@
* 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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
index 06c15cb..2419e24 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
@@ -80,9 +80,9 @@
* -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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
index 19d1bf3..09583df 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
@@ -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);
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index db78e6d..3ed776b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -494,15 +494,15 @@
* 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 @@
/* 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 @@
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 @@
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 @@
* -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_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 @@
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 @@
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_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 @@
* 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 8b93e65..a751c24 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -79,7 +79,7 @@
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,
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 717da09..1a20ca5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -47,14 +47,15 @@
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];
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 6ec818e..a11e408 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -191,7 +191,7 @@
short streamLen;
short numSamplesRead;
- short lenDecodedAudio;
+ int lenDecodedAudio;
short senderIdx;
short receiverIdx;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 959be7d..b1d8a24 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -62,7 +62,7 @@
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 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index c05d773..1eeb5ca 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -198,7 +198,7 @@
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_)),
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
index dccc7ca..925cd85 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
@@ -64,11 +64,11 @@
* 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 @@
* 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 @@
* 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 @@
* 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(...)
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index a30b1cb..328fc48 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -131,10 +131,10 @@
}
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::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.
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 527de10..e250616 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -78,11 +78,11 @@
}
}
-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 @@
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 @@
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 @@
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;