Revert "Upconvert various types to int."
This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.
BUG=499241
TBR=hlundin
Review URL: https://codereview.webrtc.org/1179953003
Cr-Commit-Position: refs/heads/master@{#9418}
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 1ec5d67..b016f40 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
*/
-int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
- int16_t quality);
+int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t 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
*/
-int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
- int16_t nrOfSamples, uint8_t* SIDdata,
- int16_t* bytesOut, int16_t forceSID);
+int16_t 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 32e2859..9862f12 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;
- int enc_sampfreq;
+ uint16_t enc_sampfreq;
int16_t enc_interval;
int16_t enc_msSinceSID;
int32_t enc_Energy;
@@ -142,8 +142,8 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
- int16_t quality) {
+int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t 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
*/
-int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
- int16_t nrOfSamples, uint8_t* SIDdata,
- int16_t* bytesOut, int16_t forceSID) {
+int16_t 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];
@@ -388,12 +388,10 @@
inst->enc_msSinceSID = 0;
*bytesOut = inst->enc_nrOfCoefs + 1;
- inst->enc_msSinceSID +=
- (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
+ inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
return inst->enc_nrOfCoefs + 1;
} else {
- inst->enc_msSinceSID +=
- (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
+ inst->enc_msSinceSID += (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 6a669e2..d06c588 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 @@
}
}
-int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
// Free encoder memory
return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
@@ -79,7 +79,7 @@
}
}
-int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+int16_t 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 a5ecbe7..7fe11a7 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
*/
-int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
@@ -142,7 +142,7 @@
* -1 - Error
*/
-int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int16_t 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 c24b4a6..d8f8c93 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) */
- int scale) /* (i) Scale factor to use for
+ int16_t 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 a0435c4..533d0a4 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -36,6 +36,7 @@
int16_t low, /* (i) Lag to start from (typically
20) */
int16_t high, /* (i) Lag to end at (typically 39 */
- int scale); /* (i) Scale factor to use for the crossDot */
+ int16_t 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 2b7e082..f8a0933 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 68dd7da..1b50c0b 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 39f18c2..7e6daf9 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 e73d414..6c181bd 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 bfe0e64..b1c0f8c 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 c7bf929..c7e1e54 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 */
- int scale, /* (i) The scaling of all energy values */
+ int16_t 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 877a1c6..a775a02 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -46,9 +46,7 @@
int16_t block /* (i) the subblock number */
) {
int16_t i, j, stage, range;
- int16_t *pp;
- int16_t tmp;
- int scale;
+ int16_t *pp, scale, tmp;
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 257013c..f282432 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -120,8 +120,8 @@
shifts = WEBRTC_SPL_MAX(0, shifts);
/* compute cross correlation */
- WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
- shifts, -1);
+ WebRtcSpl_CrossCorrelation(corr32, target, regressor,
+ ENH_BLOCKL_HALF, 50, (int16_t)shifts, -1);
/* Find 3 highest correlations that should be compared for the
highest (corr*corr)/ener */
@@ -206,8 +206,8 @@
shifts=0;
/* compute cross correlation */
- WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
- 1);
+ WebRtcSpl_CrossCorrelation(corr32, target, regressor,
+ plc_blockl, 3, (int16_t)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 e41c095..88ad33b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -88,10 +88,10 @@
}
}
-int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
- const int16_t* speechIn,
- int16_t len,
- uint8_t* encoded) {
+int16_t 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 @@
}
-int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
- const uint8_t* encoded,
- int16_t len,
- int16_t* decoded,
- int16_t* speechType)
+int16_t 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);
}
-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)
{
int i=0;
if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
@@ -222,11 +222,11 @@
return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
}
-int WebRtcIlbcfix_Decode30Ms(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 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 0659e50..d903ac7 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.
*---------------------------------------------------------------*/
-int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
+int16_t 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 cdd2192..4871b5c 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.
*---------------------------------------------------------------*/
-int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
+int16_t 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 9c562db..1a2fa08 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.
*---------------------------------------------------------------*/
-int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
+int16_t 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 7154661..2eea27c 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.
*---------------------------------------------------------------*/
-int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
+int16_t 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 4934968..b7e1735 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
*/
- int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
- const int16_t *speechIn,
- int16_t len,
- uint8_t* encoded);
+ int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
+ const int16_t *speechIn,
+ int16_t len,
+ uint8_t* encoded);
/****************************************************************************
* WebRtcIlbcfix_DecoderInit(...)
@@ -180,21 +180,21 @@
* -1 - Error
*/
- 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,
+ int16_t WebRtcIlbcfix_Decode(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(...)
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
index 0da6d54..048745a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -29,8 +29,7 @@
const int16_t *seq2, /* (i) second sequence */
int16_t dim2 /* (i) dimension seq2 */
){
- int16_t max, loops;
- int scale;
+ int16_t max, scale, loops;
/* Calculate correlation between the two sequences. Scale the
result of the multiplcication to maximum 26 bits in order
@@ -38,7 +37,7 @@
max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
scale=WebRtcSpl_GetSizeInBits(max);
- scale = 2 * scale - 26;
+ scale = (int16_t)(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 103a136..df37bec 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -41,8 +41,7 @@
{
FILE *ifileid,*efileid,*ofileid, *chfileid;
short encoded_data[55], data[240], speechType;
- int len;
- short mode, pli;
+ short len, mode, pli;
int blockcount = 0;
IlbcEncoderInstance *Enc_Inst;
@@ -178,7 +177,7 @@
/* decoding */
fprintf(stderr, "--- Decoding block %i --- ",blockcount);
if (pli==1) {
- len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, (int16_t)len, data,
+ len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data,
&speechType);
if (len < 0) {
fprintf(stderr, "Error decoding\n");
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 befb355..65c5b90 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.
- int ret =
+ int16_t 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 a1eb271..bf9f875 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,
- int framesize) {
+ int16_t framesize) {
return WebRtcIsacfix_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
- int frame_size_ms,
+ int16_t 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 int DecodeInternal(instance_type* inst,
- const uint8_t* encoded,
- int16_t len,
- int16_t* decoded,
- int16_t* speech_type) {
+ static inline int16_t 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 int Encode(instance_type* inst,
- const int16_t* speech_in,
- uint8_t* encoded) {
+ static inline int16_t 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 92dcf51..961fd3f 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
*/
- int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
- const int16_t *speechIn,
- uint8_t* encoded);
+ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+ const int16_t *speechIn,
+ uint8_t* encoded);
@@ -251,11 +251,11 @@
* -1 - Error
*/
- int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
- const uint8_t* encoded,
- int16_t len,
- int16_t *decoded,
- int16_t *speechType);
+ int16_t 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
- int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const uint16_t *encoded,
- int16_t len,
- int16_t *decoded,
- int16_t *speechType);
+ int16_t 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,
- int framesize);
+ int16_t rate,
+ int16_t framesize);
@@ -407,7 +407,7 @@
int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
int16_t rateBPS,
- int frameSizeMs,
+ int16_t 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 808aeb7..23048a5 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
*/
-int WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
- Bitstr_dec *streamData,
- const int32_t *envQ8,
- const int16_t lenData)
+int16_t 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 40bbb4c..584bc47 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
*/
-int WebRtcIsacfix_DecLogisticMulti2(
+int16_t 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 8c5c7e6..1270cc3 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);
-int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
- IsacFixDecoderInstance* ISACdec_obj,
- int16_t* current_framesamples);
+int16_t 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 f0ae07e..5e095da 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 @@
-int WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
- IsacFixDecoderInstance *ISACdec_obj,
- int16_t *current_framesamples)
+int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
+ IsacFixDecoderInstance *ISACdec_obj,
+ int16_t *current_framesamples)
{
int k;
int err;
int16_t BWno;
- int len = 0;
+ int16_t 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 aab8f43..3965378 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
*/
-int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
- int16_t *frQ7,
- int16_t *fiQ7,
- int16_t AvgPitchGain_Q12)
+int16_t 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;
- int len;
+ int16_t 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 e4489df..ec20c71 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) */
-int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
- int16_t *frQ7,
- int16_t *fiQ7,
- int16_t AvgPitchGain_Q12);
+int16_t 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 f1e5cd0..f8abc8a 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
*/
-int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
- const int16_t *speechIn,
- uint8_t* encoded)
+int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+ const int16_t *speechIn,
+ uint8_t* encoded)
{
ISACFIX_SubStruct *ISAC_inst;
- int stream_len;
+ int16_t 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 = -(int16_t)stream_len;
+ ISAC_inst->errorcode = - stream_len;
return -1;
}
@@ -766,17 +766,17 @@
*/
-int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
- const uint8_t* encoded,
- int16_t len,
- int16_t *decoded,
- int16_t *speechType)
+int16_t 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;
- int declen = 0;
+ int16_t 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 = -(int16_t)declen;
+ ISAC_inst->errorcode = -declen;
memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES);
return -1;
}
@@ -859,17 +859,17 @@
*/
#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const uint16_t *encoded,
- int16_t len,
- int16_t *decoded,
- int16_t *speechType)
+int16_t 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;
- int declen = 0;
+ int16_t declen = 0;
int16_t dummy[FRAMESAMPLES/2];
@@ -903,7 +903,7 @@
if (declen < 0) {
/* Some error inside the decoder */
- ISAC_inst->errorcode = -(int16_t)declen;
+ ISAC_inst->errorcode = -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,
- int framesize)
+ int16_t rate,
+ int16_t 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 = (int16_t)((FS/1000) * framesize);
+ ISAC_inst->ISACenc_obj.new_framelength = (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,
- int frameSizeMs,
+ int16_t 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 = (int16_t)((FS/1000) * frameSizeMs);
+ ISAC_inst->ISACenc_obj.new_framelength = (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 ac1efe4..a7a80ab 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -101,15 +101,14 @@
int i, errtype, h = 0, k, packetLossPercent = 0;
int16_t CodingMode;
int16_t bottleneck;
- int framesize = 30; /* ms */
+ int16_t 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;
- int declen = 0;
+ int16_t framecnt, declen = 0;
int16_t shortdata[FRAMESAMPLES_10ms];
int16_t decoded[MAX_FRAMESAMPLES];
uint16_t streamdata[500];
@@ -767,7 +766,7 @@
#else
declen = -1;
#endif
- prevFrameSize = static_cast<int16_t>(declen / 240);
+ prevFrameSize = 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 96688b9..b4c0ee4 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;
- int framesize = 30; /* ms */
- // int framesize = 60; /* To invoke cisco complexity case at frame 2252 */
+ int16_t framesize = 30; /* ms */
+ // int16_t 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;
- int declen;
+ int16_t 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 8c70533..5a75807 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,
- int framesize) {
+ int16_t framesize) {
return WebRtcIsac_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
- int frame_size_ms,
+ int16_t 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 int DecodeInternal(instance_type* inst,
- const uint8_t* encoded,
- int16_t len,
- int16_t* decoded,
- int16_t* speech_type) {
+ static inline int16_t 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 int Encode(instance_type* inst,
- const int16_t* speech_in,
- uint8_t* encoded) {
+ static inline int16_t 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 1a83d72..6d0c32d 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
*/
- int WebRtcIsac_Encode(
+ int16_t WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded);
@@ -214,7 +214,7 @@
* -1 - Error.
*/
- int WebRtcIsac_Decode(
+ int16_t 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,
- int framesize);
+ int16_t framesize);
/******************************************************************************
@@ -300,7 +300,7 @@
int16_t WebRtcIsac_ControlBwe(
ISACStruct* ISAC_main_inst,
int32_t rateBPS,
- int frameSizeMs,
+ int16_t frameSizeMs,
int16_t enforceFrameSize);
@@ -701,7 +701,7 @@
* Return value : >0 - number of samples in decoded vector
* -1 - Error
*/
- int WebRtcIsac_DecodeRcu(
+ int16_t 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 2419e24..06c15cb 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
*/
-int WebRtcIsac_GetCrc(const int16_t* bitstream,
- int len_bitstream_in_bytes,
- uint32_t* crc)
+int16_t WebRtcIsac_GetCrc(const int16_t* bitstream,
+ int16_t 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 09583df..19d1bf3 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
*/
-int WebRtcIsac_GetCrc(
+int16_t WebRtcIsac_GetCrc(
const int16_t* encoded,
- int no_of_word8s,
- uint32_t* crc);
+ int16_t 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 3ed776b..db78e6d 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
*/
-int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
- const int16_t* speechIn,
- uint8_t* encoded) {
+int16_t 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];
- int streamLenLB = 0;
- int streamLenUB = 0;
- int streamLen = 0;
+ int16_t streamLenLB = 0;
+ int16_t streamLenUB = 0;
+ int16_t 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 =
- (int16_t)(streamLenLB + 1 + LEN_CHECK_SUM_WORD8);
+ instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
+ LEN_CHECK_SUM_WORD8;
/* Encode upper-band. */
switch (instISAC->bandwidthKHz) {
case isac12kHz: {
@@ -1045,12 +1045,12 @@
return 0;
}
-static int Decode(ISACStruct* ISAC_main_inst,
- const uint8_t* encoded,
- int16_t lenEncodedBytes,
- int16_t* decoded,
- int16_t* speechType,
- int16_t isRCUPayload) {
+static int16_t 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];
- int numDecodedBytesLB;
- int numDecodedBytesUB;
+ int16_t numDecodedBytesLB;
+ int16_t numDecodedBytesUB;
int16_t lenEncodedLBBytes;
int16_t validChecksum = 1;
int16_t k;
@@ -1350,11 +1350,11 @@
* -1 - Error
*/
-int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
- const uint8_t* encoded,
- int16_t lenEncodedBytes,
- int16_t* decoded,
- int16_t* speechType) {
+int16_t 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 @@
-int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
- const uint8_t* encoded,
- int16_t lenEncodedBytes,
- int16_t* decoded,
- int16_t* speechType) {
+int16_t 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,
- int frameSize) {
+ int16_t frameSize) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
int16_t status;
double rateLB;
@@ -1526,7 +1526,7 @@
return -1;
}
- status = ControlLb(&instISAC->instLB, rateLB, (int16_t)frameSize);
+ status = ControlLb(&instISAC->instLB, rateLB, frameSize);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
@@ -1594,7 +1594,7 @@
*/
int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
int32_t bottleneckBPS,
- int frameSizeMs,
+ int16_t 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 =
- (int16_t)((FS / 1000) * frameSizeMs);
+ instISAC->instLB.ISACencLB_obj.new_framelength = (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 73efee1..2999036 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
@@ -80,7 +80,7 @@
WebRtcIsac_EncoderInit(isac_codec_, 0);
WebRtcIsac_DecoderInit(isac_codec_);
- int encoded_bytes;
+ int16_t 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 496e8c9..c564991 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
@@ -45,15 +45,14 @@
int i, errtype, VADusage = 0, packetLossPercent = 0;
int16_t CodingMode;
int32_t bottleneck = 0;
- int framesize = 30; /* ms */
+ int16_t framesize = 30; /* ms */
int cur_framesmpls, err;
/* Runtime statistics */
double starttime, runtime, length_file;
int16_t stream_len = 0;
- int declen = 0, declenTC = 0;
- int16_t lostFrame = 0;
+ int16_t declen = 0, lostFrame = 0, declenTC = 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 a11e408..6ec818e 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;
- int lenDecodedAudio;
+ short 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 a664e0a..8f5b4cf 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;
- int frameSize = 30; /* ms */
+ int16_t 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;
- int declen = 0;
+ int16_t declen = 0;
int16_t err;
- int cur_framesmpls;
+ int16_t 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 1eeb5ca..c05d773 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_);
- int status = WebRtcOpus_Encode(
+ int16_t 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 925cd85..dccc7ca 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
*/
-int WebRtcOpus_Encode(OpusEncInst* inst,
- const int16_t* audio_in,
- int16_t samples,
- int16_t length_encoded_buffer,
- uint8_t* encoded);
+int16_t 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
*/
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
- int16_t encoded_bytes, int16_t* decoded,
- int16_t* audio_type);
+int16_t 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
*/
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
- int number_of_lost_frames);
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+ int16_t number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DecodeFec(...)
@@ -275,9 +275,9 @@
* 0 - No FEC data in the packet
* -1 - Error
*/
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
- int16_t encoded_bytes, int16_t* decoded,
- int16_t* audio_type);
+int16_t 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 328fc48..a30b1cb 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() {
- int value = WebRtcOpus_Encode(opus_encoder_,
- &in_data_[data_pointer_],
- block_length_sample_,
- max_bytes_, &bit_stream_[0]);
+ int16_t 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;
- int value_1 = 0, value_2 = 0;
+ int16_t 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 e250616..527de10 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 @@
}
}
-int WebRtcOpus_Encode(OpusEncInst* inst,
- const int16_t* audio_in,
- int16_t samples,
- int16_t length_encoded_buffer,
- uint8_t* encoded) {
+int16_t 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;
}
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
- int16_t encoded_bytes, int16_t* decoded,
- int16_t* audio_type) {
+int16_t 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;
}
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
- int number_of_lost_frames) {
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+ int16_t number_of_lost_frames) {
int16_t audio_type = 0;
int decoded_samples;
int plc_samples;
@@ -339,9 +339,9 @@
return decoded_samples;
}
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
- int16_t encoded_bytes, int16_t* decoded,
- int16_t* audio_type) {
+int16_t 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;
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index ad7e2f9..09301df 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -273,11 +273,17 @@
int16_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
for (int i = 0; i < loop_encode; i++) {
- int bitstream_len_byte_int = WebRtcOpus_Encode(
- (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
- &audio[read_samples], frame_length, kMaxBytes, bitstream);
- ASSERT_GT(bitstream_len_byte_int, -1);
- bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
+ if (channels == 1) {
+ bitstream_len_byte = WebRtcOpus_Encode(
+ opus_mono_encoder_, &audio[read_samples],
+ frame_length, kMaxBytes, bitstream);
+ ASSERT_GT(bitstream_len_byte, -1);
+ } else {
+ bitstream_len_byte = WebRtcOpus_Encode(
+ opus_stereo_encoder_, &audio[read_samples],
+ frame_length, kMaxBytes, bitstream);
+ ASSERT_GT(bitstream_len_byte, -1);
+ }
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 99ff95a..c3f1dbb 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -163,9 +163,9 @@
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
- int ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
@@ -330,11 +330,11 @@
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
- int ret = WebRtcOpus_Decode(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
if (ret > 0)
- ret *= static_cast<int>(channels_); // Return total number of samples.
+ ret *= static_cast<int16_t>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
@@ -352,11 +352,11 @@
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
- int ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
if (ret > 0)
- ret *= static_cast<int>(channels_); // Return total number of samples.
+ ret *= static_cast<int16_t>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 289e66d..7451ae2 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -272,7 +272,7 @@
}
void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
- int16_t* factor, int increment,
+ int16_t* factor, int16_t increment,
int16_t* output) {
uint16_t factor_16b = *factor;
int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
@@ -284,7 +284,7 @@
*factor = factor_16b;
}
-void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
+void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
int32_t factor = (16384 << 6) + 32;
for (size_t i = 0; i < length; i++) {
signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.h b/webrtc/modules/audio_coding/neteq/dsp_helper.h
index f903256..af4f4d6 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.h
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -110,11 +110,11 @@
// sample and increases the gain by |increment| (Q20) for each sample. The
// result is written to |output|. |length| samples are processed.
static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
- int increment, int16_t* output);
+ int16_t increment, int16_t* output);
// Starts at unity gain and gradually fades out |signal|. For each sample,
// the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
- static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
+ static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
// Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
// has |input_length| samples, and the method will write |output_length|
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index cfd2701..1378241 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -239,12 +239,14 @@
if (consecutive_expands_ == 3) {
// Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
// mute_slope = 0.0010 / fs_mult in Q20.
- parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
+ parameters.mute_slope = std::max(parameters.mute_slope,
+ static_cast<int16_t>(1049 / fs_mult));
}
if (consecutive_expands_ == 7) {
// Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
// mute_slope = 0.0020 / fs_mult in Q20.
- parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
+ parameters.mute_slope = std::max(parameters.mute_slope,
+ static_cast<int16_t>(2097 / fs_mult));
}
// Mute segment according to slope value.
@@ -366,7 +368,7 @@
InitializeForAnExpandPeriod();
// Calculate correlation in downsampled domain (4 kHz sample rate).
- int correlation_scale;
+ int16_t correlation_scale;
int correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
// If it is decided to break bit-exactness |correlation_length| should be
// initialized to the return value of Correlation().
@@ -444,7 +446,7 @@
correlation_length + start_index + correlation_lags - 1);
correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
+ (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
- correlation_scale = std::max(0, correlation_scale);
+ correlation_scale = std::max(static_cast<int16_t>(0), correlation_scale);
// Calculate the correlation, store in |correlation_vector2|.
WebRtcSpl_CrossCorrelation(
@@ -471,7 +473,7 @@
// Calculate the correlation coefficient between the two portions of the
// signal.
- int32_t corr_coefficient;
+ int16_t corr_coefficient;
if ((energy1 > 0) && (energy2 > 0)) {
int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
@@ -480,17 +482,17 @@
// If sum is odd, add 1 to make it even.
energy1_scale += 1;
}
- int32_t scaled_energy1 = energy1 >> energy1_scale;
- int32_t scaled_energy2 = energy2 >> energy2_scale;
- int16_t sqrt_energy_product = static_cast<int16_t>(
- WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
+ int16_t scaled_energy1 = energy1 >> energy1_scale;
+ int16_t scaled_energy2 = energy2 >> energy2_scale;
+ int16_t sqrt_energy_product = WebRtcSpl_SqrtFloor(
+ scaled_energy1 * scaled_energy2);
// Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
sqrt_energy_product);
- // Cap at 1.0 in Q14.
- corr_coefficient = std::min(16384, corr_coefficient);
+ corr_coefficient = std::min(static_cast<int16_t>(16384),
+ corr_coefficient); // Cap at 1.0 in Q14.
} else {
corr_coefficient = 0;
}
@@ -511,8 +513,8 @@
if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
// Energy constraint fulfilled. Use both vectors and scale them
// accordingly.
- int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
- int32_t scaled_energy1 = scaled_energy2 - 13;
+ int16_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+ int16_t scaled_energy1 = scaled_energy2 - 13;
// Calculate scaled_energy1 / scaled_energy2 in Q13.
int32_t energy_ratio = WebRtcSpl_DivW32W16(
WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
@@ -680,8 +682,7 @@
// voice_mix_factor = 0;
if (corr_coefficient > 7875) {
int16_t x1, x2, x3;
- // |corr_coefficient| is in Q14.
- x1 = static_cast<int16_t>(corr_coefficient);
+ x1 = corr_coefficient; // |corr_coefficient| is in Q14.
x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
x3 = (x1 * x2) >> 14;
static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
@@ -708,8 +709,8 @@
// the division.
// Shift the denominator from Q13 to Q5 before the division. The result of
// the division will then be in Q20.
- int temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
- (distortion_lag * slope) >> 8);
+ int16_t temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
+ (distortion_lag * slope) >> 8);
if (slope > 14746) {
// slope > 1.8.
// Divide by 2, with proper rounding.
@@ -728,7 +729,8 @@
// Make sure the mute factor decreases from 1.0 to 0.9 in no more than
// 6.25 ms.
// mute_slope >= 0.005 / fs_mult in Q20.
- parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
+ parameters.mute_slope = std::max(static_cast<int16_t>(5243 / fs_mult),
+ parameters.mute_slope);
} else if (slope > 8028) {
parameters.mute_slope = 0;
}
@@ -750,7 +752,7 @@
}
int16_t Expand::Correlation(const int16_t* input, size_t input_length,
- int16_t* output, int* output_scale) const {
+ int16_t* output, int16_t* output_scale) const {
// Set parameters depending on sample rate.
const int16_t* filter_coefficients;
int16_t num_coefficients;
@@ -839,7 +841,7 @@
// TODO(turajs): This can be moved to BackgroundNoise class.
void Expand::GenerateBackgroundNoise(int16_t* random_vector,
size_t channel,
- int mute_slope,
+ int16_t mute_slope,
bool too_many_expands,
size_t num_noise_samples,
int16_t* buffer) {
@@ -882,7 +884,7 @@
bgn_mute_factor > 0) {
// Fade BGN to zero.
// Calculate muting slope, approximately -2^18 / fs_hz.
- int mute_slope;
+ int16_t mute_slope;
if (fs_hz_ == 8000) {
mute_slope = -32;
} else if (fs_hz_ == 16000) {
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 672ea39..5679ec1 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -72,7 +72,7 @@
void GenerateBackgroundNoise(int16_t* random_vector,
size_t channel,
- int mute_slope,
+ int16_t mute_slope,
bool too_many_expands,
size_t num_noise_samples,
int16_t* buffer);
@@ -113,7 +113,7 @@
AudioVector expand_vector0;
AudioVector expand_vector1;
bool onset;
- int mute_slope; /* Q20 */
+ int16_t mute_slope; /* Q20 */
};
// Calculate the auto-correlation of |input|, with length |input_length|
@@ -121,7 +121,7 @@
// |input|, and is written to |output|. The scale factor is written to
// |output_scale|. Returns the length of the correlation vector.
int16_t Correlation(const int16_t* input, size_t input_length,
- int16_t* output, int* output_scale) const;
+ int16_t* output, int16_t* output_scale) const;
void UpdateLagIndex();
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index 23382ac..44fc511 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -311,7 +311,7 @@
const int max_corr_length = kMaxCorrelationLength;
int stop_position_downsamp = std::min(
max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
- int correlation_shift = 0;
+ int16_t correlation_shift = 0;
if (expanded_max * input_max > 26843546) {
correlation_shift = 3;
}
@@ -330,7 +330,7 @@
int16_t* correlation_ptr = &correlation16[pad_length];
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
stop_position_downsamp);
- int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+ int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
correlation, norm_shift);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 2d4ff27..6512515 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -1271,7 +1271,7 @@
*operation == kPreemptiveExpand);
packet_list->pop_front();
size_t payload_length = packet->payload_length;
- int decode_length;
+ int16_t decode_length;
if (packet->sync_packet) {
// Decode to silence with the same frame size as the last decode.
LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index b172d56..18ba79b 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -110,7 +110,7 @@
}
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int increment = static_cast<int>(64 / fs_mult);
+ int16_t increment = 64 / fs_mult;
for (size_t i = 0; i < length_per_channel; i++) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
@@ -176,7 +176,7 @@
// Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
// still ramping up from previous muting.
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int increment = static_cast<int>(64 / fs_mult);
+ int16_t increment = 64 / fs_mult;
size_t length_per_channel = length / output->Channels();
for (size_t i = 0; i < length_per_channel; i++) {
for (size_t channel_ix = 0; channel_ix < output->Channels();
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 1ef9ce5..c097f5f 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -1561,7 +1561,7 @@
int useVAD,
int bitrate,
int numChannels) {
- int cdlen = 0;
+ short cdlen = 0;
int16_t* tempdata;
static int first_cng = 1;
int16_t tempLen;