Rename _t struct types in audio_coding.

_t names are reserved in POSIX.

R=henrik.lundin@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 614a3df..9862f12 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -16,7 +16,7 @@
 #include "cng_helpfuns.h"
 #include "signal_processing_library.h"
 
-typedef struct WebRtcCngDecInst_t_ {
+typedef struct WebRtcCngDecoder_ {
   uint32_t dec_seed;
   int32_t dec_target_energy;
   int32_t dec_used_energy;
@@ -32,9 +32,9 @@
   int16_t target_scale_factor;  /* Q13 */
   int16_t errorcode;
   int16_t initflag;
-} WebRtcCngDecInst_t;
+} WebRtcCngDecoder;
 
-typedef struct WebRtcCngEncInst_t_ {
+typedef struct WebRtcCngEncoder_ {
   int16_t enc_nrOfCoefs;
   uint16_t enc_sampfreq;
   int16_t enc_interval;
@@ -45,7 +45,7 @@
   uint32_t enc_seed;
   int16_t errorcode;
   int16_t initflag;
-} WebRtcCngEncInst_t;
+} WebRtcCngEncoder;
 
 const int32_t WebRtcCng_kDbov[94] = {
   1081109975,  858756178,  682134279,  541838517,  430397633,  341876992,
@@ -84,10 +84,10 @@
  */
 int16_t WebRtcCng_CreateEnc(CNG_enc_inst** cng_inst) {
   if (cng_inst != NULL) {
-    *cng_inst = (CNG_enc_inst*) malloc(sizeof(WebRtcCngEncInst_t));
+    *cng_inst = (CNG_enc_inst*) malloc(sizeof(WebRtcCngEncoder));
     if (*cng_inst != NULL) {
-      (*(WebRtcCngEncInst_t**) cng_inst)->errorcode = 0;
-      (*(WebRtcCngEncInst_t**) cng_inst)->initflag = 0;
+      (*(WebRtcCngEncoder**) cng_inst)->errorcode = 0;
+      (*(WebRtcCngEncoder**) cng_inst)->initflag = 0;
 
       /* Needed to get the right function pointers in SPLIB. */
       WebRtcSpl_Init();
@@ -105,10 +105,10 @@
 
 int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
   if (cng_inst != NULL ) {
-    *cng_inst = (CNG_dec_inst*) malloc(sizeof(WebRtcCngDecInst_t));
+    *cng_inst = (CNG_dec_inst*) malloc(sizeof(WebRtcCngDecoder));
     if (*cng_inst != NULL ) {
-      (*(WebRtcCngDecInst_t**) cng_inst)->errorcode = 0;
-      (*(WebRtcCngDecInst_t**) cng_inst)->initflag = 0;
+      (*(WebRtcCngDecoder**) cng_inst)->errorcode = 0;
+      (*(WebRtcCngDecoder**) cng_inst)->initflag = 0;
 
       /* Needed to get the right function pointers in SPLIB. */
       WebRtcSpl_Init();
@@ -145,8 +145,8 @@
 int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
                           int16_t quality) {
   int i;
-  WebRtcCngEncInst_t* inst = (WebRtcCngEncInst_t*) cng_inst;
-  memset(inst, 0, sizeof(WebRtcCngEncInst_t));
+  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
+  memset(inst, 0, sizeof(WebRtcCngEncoder));
 
   /* Check LPC order */
   if (quality > WEBRTC_CNG_MAX_LPC_ORDER || quality <= 0) {
@@ -172,9 +172,9 @@
 int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
   int i;
 
-  WebRtcCngDecInst_t* inst = (WebRtcCngDecInst_t*) cng_inst;
+  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
 
-  memset(inst, 0, sizeof(WebRtcCngDecInst_t));
+  memset(inst, 0, sizeof(WebRtcCngDecoder));
   inst->dec_seed = 7777;  /* For debugging only. */
   inst->dec_order = 5;
   inst->dec_target_scale_factor = 0;
@@ -230,7 +230,7 @@
 int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
                          int16_t nrOfSamples, uint8_t* SIDdata,
                          int16_t* bytesOut, int16_t forceSID) {
-  WebRtcCngEncInst_t* inst = (WebRtcCngEncInst_t*) cng_inst;
+  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
 
   int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
   int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@@ -413,7 +413,7 @@
 int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
                             size_t length) {
 
-  WebRtcCngDecInst_t* inst = (WebRtcCngDecInst_t*) cng_inst;
+  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
   int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
   int32_t targetEnergy;
   int i;
@@ -474,7 +474,7 @@
  */
 int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
                            int16_t nrOfSamples, int16_t new_period) {
-  WebRtcCngDecInst_t* inst = (WebRtcCngDecInst_t*) cng_inst;
+  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
 
   int i;
   int16_t excitation[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
@@ -591,12 +591,12 @@
  */
 int16_t WebRtcCng_GetErrorCodeEnc(CNG_enc_inst* cng_inst) {
   /* Typecast pointer to real structure. */
-  WebRtcCngEncInst_t* inst = (WebRtcCngEncInst_t*) cng_inst;
+  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
   return inst->errorcode;
 }
 
 int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst) {
   /* Typecast pointer to real structure. */
-  WebRtcCngDecInst_t* inst = (WebRtcCngDecInst_t*) cng_inst;
+  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
   return inst->errorcode;
 }
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
index 65ca77b..ee0eb89 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
@@ -62,9 +62,9 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-static void block4(g722_decode_state_t *s, int band, int d);
+static void block4(G722DecoderState *s, int band, int d);
 
-static void block4(g722_decode_state_t *s, int band, int d)
+static void block4(G722DecoderState *s, int band, int d)
 {
     int wd1;
     int wd2;
@@ -154,13 +154,12 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
-                                             int rate,
-                                             int options)
-{
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+                                          int rate,
+                                          int options) {
     if (s == NULL)
     {
-        if ((s = (g722_decode_state_t *) malloc(sizeof(*s))) == NULL)
+        if ((s = (G722DecoderState *) malloc(sizeof(*s))) == NULL)
             return NULL;
     }
     memset(s, 0, sizeof(*s));
@@ -182,14 +181,14 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-int WebRtc_g722_decode_release(g722_decode_state_t *s)
+int WebRtc_g722_decode_release(G722DecoderState *s)
 {
     free(s);
     return 0;
 }
 /*- End of function --------------------------------------------------------*/
 
-int WebRtc_g722_decode(g722_decode_state_t *s, int16_t amp[],
+int WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
                        const uint8_t g722_data[], int len)
 {
     static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h b/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
index da61224..5cd1b2d 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -91,7 +91,7 @@
     int in_bits;
     unsigned int out_buffer;
     int out_bits;
-} g722_encode_state_t;
+} G722EncoderState;
 
 typedef struct
 {
@@ -129,26 +129,26 @@
     int in_bits;
     unsigned int out_buffer;
     int out_bits;
-} g722_decode_state_t;
+} G722DecoderState;
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
-                                             int rate,
-                                             int options);
-int WebRtc_g722_encode_release(g722_encode_state_t *s);
-int WebRtc_g722_encode(g722_encode_state_t *s,
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+                                          int rate,
+                                          int options);
+int WebRtc_g722_encode_release(G722EncoderState *s);
+int WebRtc_g722_encode(G722EncoderState *s,
                        uint8_t g722_data[],
                        const int16_t amp[],
                        int len);
 
-g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
-                                             int rate,
-                                             int options);
-int WebRtc_g722_decode_release(g722_decode_state_t *s);
-int WebRtc_g722_decode(g722_decode_state_t *s,
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+                                          int rate,
+                                          int options);
+int WebRtc_g722_decode_release(G722DecoderState *s);
+int WebRtc_g722_decode(G722DecoderState *s,
                        int16_t amp[],
                        const uint8_t g722_data[],
                        int len);
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_encode.c b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
index 03ac3c5..bed2d21 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
@@ -62,7 +62,7 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-static void block4(g722_encode_state_t *s, int band, int d)
+static void block4(G722EncoderState *s, int band, int d)
 {
     int wd1;
     int wd2;
@@ -151,12 +151,12 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
-                                             int rate, int options)
-{
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+                                          int rate,
+                                          int options) {
     if (s == NULL)
     {
-        if ((s = (g722_encode_state_t *) malloc(sizeof(*s))) == NULL)
+        if ((s = (G722EncoderState *) malloc(sizeof(*s))) == NULL)
             return NULL;
     }
     memset(s, 0, sizeof(*s));
@@ -178,7 +178,7 @@
 }
 /*- End of function --------------------------------------------------------*/
 
-int WebRtc_g722_encode_release(g722_encode_state_t *s)
+int WebRtc_g722_encode_release(G722EncoderState *s)
 {
     free(s);
     return 0;
@@ -202,7 +202,7 @@
 }
 #endif
 
-int WebRtc_g722_encode(g722_encode_state_t *s, uint8_t g722_data[],
+int WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
                        const int16_t amp[], int len)
 {
     static const int q6[32] =
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index a2155e8..9a67d88 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -18,7 +18,7 @@
 
 int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
 {
-    *G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
+    *G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
     if (*G722enc_inst!=NULL) {
       return(0);
     } else {
@@ -31,7 +31,7 @@
     // Create and/or reset the G.722 encoder
     // Bitrate 64 kbps and wideband mode (2)
     G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
-        (g722_encode_state_t*) G722enc_inst, 64000, 2);
+        (G722EncoderState*) G722enc_inst, 64000, 2);
     if (G722enc_inst == NULL) {
         return -1;
     } else {
@@ -42,7 +42,7 @@
 int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
 {
     // Free encoder memory
-    return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
+    return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
 }
 
 int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
@@ -52,13 +52,13 @@
 {
     unsigned char *codechar = (unsigned char*) encoded;
     // Encode the input speech vector
-    return WebRtc_g722_encode((g722_encode_state_t*) G722enc_inst,
+    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst,
                        codechar, speechIn, len);
 }
 
 int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
 {
-    *G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
+    *G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
     if (*G722dec_inst!=NULL) {
       return(0);
     } else {
@@ -71,7 +71,7 @@
     // Create and/or reset the G.722 decoder
     // Bitrate 64 kbps and wideband mode (2)
     G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
-        (g722_decode_state_t*) G722dec_inst, 64000, 2);
+        (G722DecoderState*) G722dec_inst, 64000, 2);
     if (G722dec_inst == NULL) {
         return -1;
     } else {
@@ -82,7 +82,7 @@
 int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
 {
     // Free encoder memory
-    return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
+    return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
 }
 
 int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
@@ -93,7 +93,7 @@
 {
     // Decode the G.722 encoder stream
     *speechType=G722_WEBRTC_SPEECH;
-    return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
+    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst,
                               decoded, (uint8_t*) encoded, len);
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
index 8a8b133..75fc970 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -27,7 +27,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_AbsQuant(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
                                    and idxVec, uses state_first as
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
index c061ff4..5154534 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -27,7 +27,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_AbsQuant(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
                                    and idxVec, uses state_first as
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
index 667240a..8a89c87 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -34,7 +34,7 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CbSearch(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) the encoder state structure */
     int16_t *index,  /* (o) Codebook indices */
     int16_t *gain_index, /* (o) Gain quantization indices */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
index fc62190..2fe236f 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -20,7 +20,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
 
 void WebRtcIlbcfix_CbSearch(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) the encoder state structure */
     int16_t *index,  /* (o) Codebook indices */
     int16_t *gain_index, /* (o) Gain quantization indices */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
index febd4ce..3a2e5a2 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -39,7 +39,7 @@
 void WebRtcIlbcfix_DecodeImpl(
     int16_t *decblock,    /* (o) decoded signal block */
     const uint16_t *bytes, /* (i) encoded signal bits */
-    iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+    IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state
                                            structure */
     int16_t mode      /* (i) 0: bad packet, PLC,
                                                                    1: normal */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.h b/webrtc/modules/audio_coding/codecs/ilbc/decode.h
index 71e3802..0c4c2ef 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.h
@@ -28,7 +28,7 @@
 void WebRtcIlbcfix_DecodeImpl(
     int16_t *decblock,    /* (o) decoded signal block */
     const uint16_t *bytes, /* (i) encoded signal bits */
-    iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+    IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state
                                            structure */
     int16_t mode      /* (i) 0: bad packet, PLC,
                                                                    1: normal */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
index c65856b..169218a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -33,7 +33,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecodeResidual(
-    iLBC_Dec_Inst_t *iLBCdec_inst,
+    IlbcDecoder *iLBCdec_inst,
     /* (i/o) the decoder state structure */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
                                 for the decoding  */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
index 04d6227..67f05a5 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_DecodeResidual(
-    iLBC_Dec_Inst_t *iLBCdec_inst,
+    IlbcDecoder *iLBCdec_inst,
     /* (i/o) the decoder state structure */
     iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
                                    for the decoding  */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
index 3557665..fad8170 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -31,7 +31,7 @@
                                    coefficients */
     int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
     int16_t length,   /* (i) length of lsf coefficient vector */
-    iLBC_Dec_Inst_t *iLBCdec_inst
+    IlbcDecoder *iLBCdec_inst
     /* (i) the decoder state structure */
                                           ){
   int  i, pos, lp_length;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
index 02c653a..37ecf07 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -31,7 +31,7 @@
                                    coefficients */
     int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
     int16_t length,   /* (i) length of lsf coefficient vector */
-    iLBC_Dec_Inst_t *iLBCdec_inst
+    IlbcDecoder *iLBCdec_inst
     /* (i) the decoder state structure */
                                           );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/defines.h b/webrtc/modules/audio_coding/codecs/ilbc/defines.h
index 796d7b5..2d37e52 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/defines.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/defines.h
@@ -129,7 +129,7 @@
 } iLBC_bits;
 
 /* type definition encoder instance */
-typedef struct iLBC_Enc_Inst_t_ {
+typedef struct IlbcEncoder_ {
 
   /* flag for frame size mode */
   int16_t mode;
@@ -167,10 +167,10 @@
   int16_t diff;
 #endif
 
-} iLBC_Enc_Inst_t;
+} IlbcEncoder;
 
 /* type definition decoder instance */
-typedef struct iLBC_Dec_Inst_t_ {
+typedef struct IlbcDecoder_ {
 
   /* flag for frame size mode */
   int16_t mode;
@@ -214,6 +214,6 @@
   int16_t enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
   int16_t enh_period[ENH_NBLOCKS_TOT];
 
-} iLBC_Dec_Inst_t;
+} IlbcDecoder;
 
 #endif
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
index 4d233e3..53a4e9a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -34,7 +34,7 @@
     int16_t *decresidual,  /* (i) decoded residual */
     int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
     int16_t inlag,    /* (i) pitch lag */
-    iLBC_Dec_Inst_t *iLBCdec_inst
+    IlbcDecoder *iLBCdec_inst
     /* (i/o) decoder instance */
                             ){
   int16_t i, pick;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
index 15e68ec..c55b815 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -34,7 +34,7 @@
     int16_t *decresidual,  /* (i) decoded residual */
     int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
     int16_t inlag,    /* (i) pitch lag */
-    iLBC_Dec_Inst_t *iLBCdec_inst
+    IlbcDecoder *iLBCdec_inst
     /* (i/o) decoder instance */
                             );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
index 318c27a..6f7c04e 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -45,7 +45,7 @@
 void WebRtcIlbcfix_EncodeImpl(
     uint16_t *bytes,     /* (o) encoded data bits iLBC */
     const int16_t *block, /* (i) speech vector to encode */
-    iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
                                      state */
                           ){
   int n, meml_gotten, Nfor, Nback;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.h b/webrtc/modules/audio_coding/codecs/ilbc/encode.h
index 5a47c58..fe6ae62 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/encode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.h
@@ -28,7 +28,7 @@
 void WebRtcIlbcfix_EncodeImpl(
     uint16_t *bytes,     /* (o) encoded data bits iLBC */
     const int16_t *block, /* (i) speech vector to encode */
-    iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
                                            state */
                           );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index 0778c27..4f27988 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -33,7 +33,7 @@
 int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     int16_t *out,     /* (o) enhanced signal */
     int16_t *in,      /* (i) unenhanced signal */
-    iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
+    IlbcDecoder *iLBCdec_inst /* (i) buffers etc */
                                         ){
   int iblock;
   int lag=20, tlag=20;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
index f6ebed6..fa58b7a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -28,7 +28,7 @@
 int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     int16_t *out,     /* (o) enhanced signal */
     int16_t *in,      /* (i) unenhanced signal */
-    iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
+    IlbcDecoder *iLBCdec_inst /* (i) buffers etc */
                                         );
 
 #endif
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
index c2c6f75..c93cdd4 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -25,7 +25,7 @@
 
 int16_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i/o) the encoder state structure */
     int16_t *residualFIX /* (i) lpc residual signal */
                                                 ){
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
index ba9b9fe..b32e2c8 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -21,7 +21,7 @@
 
 int16_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i/o) the encoder state structure */
     int16_t *residualFIX /* (i) lpc residual signal */
                                                 );
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index 5af52bb..4d815fa 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -28,7 +28,7 @@
                                     int16_t* ILBCENC_inst_Addr,
                                     int16_t* size) {
   *iLBC_encinst=(IlbcEncoderInstance*)ILBCENC_inst_Addr;
-  *size=sizeof(iLBC_Enc_Inst_t)/sizeof(int16_t);
+  *size=sizeof(IlbcEncoder)/sizeof(int16_t);
   if (*iLBC_encinst!=NULL) {
     return(0);
   } else {
@@ -40,7 +40,7 @@
                                     int16_t* ILBCDEC_inst_Addr,
                                     int16_t* size) {
   *iLBC_decinst=(IlbcDecoderInstance*)ILBCDEC_inst_Addr;
-  *size=sizeof(iLBC_Dec_Inst_t)/sizeof(int16_t);
+  *size=sizeof(IlbcDecoder)/sizeof(int16_t);
   if (*iLBC_decinst!=NULL) {
     return(0);
   } else {
@@ -49,7 +49,7 @@
 }
 
 int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst) {
-  *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(iLBC_Enc_Inst_t));
+  *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(IlbcEncoder));
   if (*iLBC_encinst!=NULL) {
     WebRtcSpl_Init();
     return(0);
@@ -59,7 +59,7 @@
 }
 
 int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst) {
-  *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(iLBC_Dec_Inst_t));
+  *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(IlbcDecoder));
   if (*iLBC_decinst!=NULL) {
     WebRtcSpl_Init();
     return(0);
@@ -81,7 +81,7 @@
 int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
                                   int16_t mode) {
   if ((mode==20)||(mode==30)) {
-    WebRtcIlbcfix_InitEncode((iLBC_Enc_Inst_t*) iLBCenc_inst, mode);
+    WebRtcIlbcfix_InitEncode((IlbcEncoder*) iLBCenc_inst, mode);
     return(0);
   } else {
     return(-1);
@@ -95,12 +95,12 @@
   int16_t pos = 0;
   int16_t encpos = 0;
 
-  if ((len != ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
+  if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) &&
 #ifdef SPLIT_10MS
       (len != 80) &&
 #endif
-      (len != 2*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
-      (len != 3*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl))
+      (len != 2*((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+      (len != 3*((IlbcEncoder*)iLBCenc_inst)->blockl))
   {
     /* A maximum of 3 frames/packet is allowed */
     return(-1);
@@ -109,14 +109,14 @@
     /* call encoder */
     while (pos<len) {
       WebRtcIlbcfix_EncodeImpl((uint16_t*)&encoded[2 * encpos], &speechIn[pos],
-                               (iLBC_Enc_Inst_t*)iLBCenc_inst);
+                               (IlbcEncoder*)iLBCenc_inst);
 #ifdef SPLIT_10MS
       pos += 80;
-      if(((iLBC_Enc_Inst_t*)iLBCenc_inst)->section == 0)
+      if(((IlbcEncoder*)iLBCenc_inst)->section == 0)
 #else
-        pos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl;
+        pos += ((IlbcEncoder*)iLBCenc_inst)->blockl;
 #endif
-      encpos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->no_of_words;
+      encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words;
     }
     return (encpos*2);
   }
@@ -125,18 +125,18 @@
 int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
                                   int16_t mode) {
   if ((mode==20)||(mode==30)) {
-    WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, mode, 1);
+    WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, mode, 1);
     return(0);
   } else {
     return(-1);
   }
 }
 int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst) {
-  WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 20, 1);
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
   return(0);
 }
 int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
-  WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 30, 1);
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
   return(0);
 }
 
@@ -150,19 +150,19 @@
   int i=0;
   /* Allow for automatic switching between the frame sizes
      (although you do get some discontinuity) */
-  if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
     /* ok, do nothing */
   } else {
     /* Test if the mode has changed */
-    if (((iLBC_Dec_Inst_t*)iLBCdec_inst)->mode==20) {
+    if (((IlbcDecoder*)iLBCdec_inst)->mode==20) {
       if ((len==NO_OF_BYTES_30MS)||
           (len==2*NO_OF_BYTES_30MS)||
           (len==3*NO_OF_BYTES_30MS)) {
         WebRtcIlbcfix_InitDecode(
-            ((iLBC_Dec_Inst_t*)iLBCdec_inst), 30,
-            ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
+            ((IlbcDecoder*)iLBCdec_inst), 30,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
       } else {
         /* Unsupported frame length */
         return(-1);
@@ -172,8 +172,8 @@
           (len==2*NO_OF_BYTES_20MS)||
           (len==3*NO_OF_BYTES_20MS)) {
         WebRtcIlbcfix_InitDecode(
-            ((iLBC_Dec_Inst_t*)iLBCdec_inst), 20,
-            ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
+            ((IlbcDecoder*)iLBCdec_inst), 20,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
       } else {
         /* Unsupported frame length */
         return(-1);
@@ -181,17 +181,17 @@
     }
   }
 
-  while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
     WebRtcIlbcfix_DecodeImpl(
-        &decoded[i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl],
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
         (const uint16_t*)&encoded
-            [i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words],
-        (iLBC_Dec_Inst_t*)iLBCdec_inst, 1);
+            [i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
 int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance *iLBCdec_inst,
@@ -201,25 +201,25 @@
                                  int16_t *speechType)
 {
   int i=0;
-  if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
     /* ok, do nothing */
   } else {
     return(-1);
   }
 
-  while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
     WebRtcIlbcfix_DecodeImpl(
-        &decoded[i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl],
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
         (const uint16_t*)&encoded
-            [i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words],
-        (iLBC_Dec_Inst_t*)iLBCdec_inst, 1);
+            [i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
 int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance *iLBCdec_inst,
@@ -229,25 +229,25 @@
                                  int16_t *speechType)
 {
   int i=0;
-  if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
-      (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
     /* ok, do nothing */
   } else {
     return(-1);
   }
 
-  while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
     WebRtcIlbcfix_DecodeImpl(
-        &decoded[i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl],
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
         (const uint16_t*)&encoded
-            [i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words],
-        (iLBC_Dec_Inst_t*)iLBCdec_inst, 1);
+            [i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1);
     i++;
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
 int16_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
@@ -259,10 +259,10 @@
   for (i=0;i<noOfLostFrames;i++) {
     /* call decoder */
     WebRtcIlbcfix_DecodeImpl(
-        &decoded[i * ((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], &dummy,
-        (iLBC_Dec_Inst_t*)iLBCdec_inst, 0);
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], &dummy,
+        (IlbcDecoder*)iLBCdec_inst, 0);
   }
-  return (noOfLostFrames*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+  return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
 int16_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
@@ -272,8 +272,8 @@
   (void)(decoded = NULL);
   (void)(noOfLostFrames = 0);
 
-  WebRtcSpl_MemSetW16(((iLBC_Dec_Inst_t*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
-  ((iLBC_Dec_Inst_t*)iLBCdec_inst)->prev_enh_pl = 2;
+  WebRtcSpl_MemSetW16(((IlbcDecoder*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
+  ((IlbcDecoder*)iLBCdec_inst)->prev_enh_pl = 2;
 
   return (0);
 }
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
index f76ade0..d903ac7 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -24,7 +24,7 @@
  *---------------------------------------------------------------*/
 
 int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
-    iLBC_Dec_Inst_t *iLBCdec_inst,  /* (i/o) Decoder instance */
+    IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
     int16_t mode,  /* (i) frame size mode */
     int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
   int i;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
index 2938c74..4871b5c 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
-    iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
     int16_t mode,     /* (i) frame size mode */
     int use_enhancer           /* (i) 1 to use enhancer
                                   0 to run without enhancer */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
index f6da158..1a2fa08 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -24,7 +24,7 @@
  *---------------------------------------------------------------*/
 
 int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
-    iLBC_Enc_Inst_t *iLBCenc_inst,  /* (i/o) Encoder instance */
+    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 562efb1..2eea27c 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
-    iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+    IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
     int16_t mode     /* (i) frame size mode */
                                          );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
index 8f9a3c0..258b6b1 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -34,7 +34,7 @@
                                    before/after encoding */
     int16_t *lsf_index,  /* (o) lsf quantization index */
     int16_t *data,   /* (i) Speech to do LPC analysis on */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                               ) {
   /* Stack based */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
index 8d87b0a..9f6f504 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -32,7 +32,7 @@
                                    before/after encoding */
     int16_t *lsf_index,  /* (o) lsf quantization index */
     int16_t *data,   /* (i) Speech to do LPC analysis on */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                              );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
index 9055493..d89770e 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -39,7 +39,7 @@
     int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
                                    previous signal frame Q13 */
     int16_t length,  /* (i) should equate FILTERORDER */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         ) {
   int i, pos, lp_length;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
index b11f5d8..6abcc00 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -39,7 +39,7 @@
     int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
                                    previous signal frame Q13 */
     int16_t length,  /* (i) should equate FILTERORDER */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
index f8125e1..dfc637b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -29,7 +29,7 @@
 void WebRtcIlbcfix_SimpleLpcAnalysis(
     int16_t *lsf,   /* (o) lsf coefficients */
     int16_t *data,   /* (i) new block of speech */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                      ) {
   int k;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
index a3d1985..7808da1 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -28,7 +28,7 @@
 void WebRtcIlbcfix_SimpleLpcAnalysis(
     int16_t *lsf,   /* (o) lsf coefficients */
     int16_t *data,   /* (i) new block of speech */
-    iLBC_Enc_Inst_t *iLBCenc_inst
+    IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                      );
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_search.c b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
index bf9c7a9..5d85a84 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -25,7 +25,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateSearch(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
                                and idxVec, input state_first) */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_search.h b/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
index 80a4f13..800beac 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
@@ -26,7 +26,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateSearch(
-    iLBC_Enc_Inst_t *iLBCenc_inst,
+    IlbcEncoder *iLBCenc_inst,
     /* (i) Encoder instance */
     iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
                                and idxVec, input state_first) */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
index 934d4be..303ede3 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -40,7 +40,7 @@
  *---------------------------------------------------------------*/
 
 short encode(                         /* (o) Number of bytes encoded */
-    iLBC_Enc_Inst_t *iLBCenc_inst,    /* (i/o) Encoder instance */
+    IlbcEncoder *iLBCenc_inst,    /* (i/o) Encoder instance */
     int16_t *encoded_data,      /* (o) The encoded bytes */
     int16_t *data               /* (i) The signal block to encode */
                                                         ){
@@ -56,7 +56,7 @@
  *---------------------------------------------------------------*/
 
 short decode( /* (o) Number of decoded samples */
-    iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
     short *decoded_data, /* (o) Decoded signal block */
     short *encoded_data, /* (i) Encoded bytes */
     short mode           /* (i) 0=PL, 1=Normal */
@@ -100,8 +100,8 @@
   short *channeldata;
   int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
   short mode;
-  iLBC_Enc_Inst_t Enc_Inst;
-  iLBC_Dec_Inst_t Dec_Inst;
+  IlbcEncoder Enc_Inst;
+  IlbcDecoder Dec_Inst;
 
   short frameLen;
   short count;
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 a38c6e5..488ee2e 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -33,19 +33,19 @@
                                     uint32_t arr_ts);
 
 int16_t WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
-                                       ISACFIX_DecInst_t* ISACdec_obj,
+                                       IsacFixDecoderInstance* ISACdec_obj,
                                        int16_t* current_framesamples);
 
 int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
-                                          ISACFIX_DecInst_t* ISACdec_obj,
+                                          IsacFixDecoderInstance* ISACdec_obj,
                                           int16_t* current_framesample );
 
 int WebRtcIsacfix_EncodeImpl(int16_t* in,
-                             ISACFIX_EncInst_t* ISACenc_obj,
+                             IsacFixEncoderInstance* ISACenc_obj,
                              BwEstimatorstr* bw_estimatordata,
                              int16_t CodingMode);
 
-int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t* ISACenc_obj,
+int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance* ISACenc_obj,
                                    int BWnumber,
                                    float scale);
 
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 fb9c7be..3126f55 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -28,7 +28,7 @@
 
 
 int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
-                                 ISACFIX_DecInst_t *ISACdec_obj,
+                                 IsacFixDecoderInstance *ISACdec_obj,
                                  int16_t       *current_framesamples)
 {
   int k;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
index 51344c2..510f5e8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
@@ -236,7 +236,7 @@
 
 
 int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
-                                    ISACFIX_DecInst_t *ISACdec_obj,
+                                    IsacFixDecoderInstance *ISACdec_obj,
                                     int16_t *current_framesamples )
 {
   int subframecnt;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
index 2c63c02..4090720 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -32,7 +32,7 @@
 
 
 int WebRtcIsacfix_EncodeImpl(int16_t      *in,
-                             ISACFIX_EncInst_t  *ISACenc_obj,
+                             IsacFixEncoderInstance  *ISACenc_obj,
                              BwEstimatorstr      *bw_estimatordata,
                              int16_t         CodingMode)
 {
@@ -491,7 +491,7 @@
    The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl()
    is used. The data needed is taken from the struct, where it was stored
    when calling the encoder. */
-int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t  *ISACenc_obj,
+int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance  *ISACenc_obj,
                                    int     BWnumber,
                                    float              scale)
 {
@@ -503,7 +503,7 @@
   int16_t model;
   const uint16_t *Q_PitchGain_cdf_ptr[1];
   const uint16_t **cdf;
-  const ISAC_SaveEncData_t *SaveEnc_str;
+  const IsacSaveEncoderData *SaveEnc_str;
   int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
   int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1];
   int16_t tmp_fre[FRAMESAMPLES];
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 869db75..c80de04 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
@@ -1059,7 +1059,7 @@
                           int16_t *model,
                           int32_t *sizeQ11,
                           Bitstr_enc *streamdata,
-                          ISAC_SaveEncData_t* encData,
+                          IsacSaveEncoderData* encData,
                           transcode_obj *transcodingParam) {
   int j, k, n;
   int16_t posQQ, pos2QQ, gainpos;
@@ -1334,7 +1334,7 @@
 
 int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17,
                                  Bitstr_enc *streamdata,
-                                 ISAC_SaveEncData_t* encData) {
+                                 IsacSaveEncoderData* encData) {
   int j, k;
   int16_t posQQ, pos2QQ, gainpos;
   int16_t posg;
@@ -1440,7 +1440,7 @@
                             int16_t *model,
                             int32_t *sizeQ11,
                             Bitstr_enc *streamdata,
-                            ISAC_SaveEncData_t* encData,
+                            IsacSaveEncoderData* encData,
                             transcode_obj *transcodeParam)
 {
   int status = 0;
@@ -1599,8 +1599,9 @@
 
 
 /* quantize & code Pitch Gains */
-int WebRtcIsacfix_EncodePitchGain(int16_t *PitchGains_Q12, Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
-{
+int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
+                                  Bitstr_enc* streamdata,
+                                  IsacSaveEncoderData* encData) {
   int k,j;
   int16_t SQ15[PITCH_SUBFRAMES];
   int16_t index[3];
@@ -1754,9 +1755,10 @@
 
 
 /* quantize & code Pitch Lags */
-int WebRtcIsacfix_EncodePitchLag(int16_t *PitchLagsQ7,int16_t *PitchGain_Q12,
-                                 Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
-{
+int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
+                                 int16_t* PitchGain_Q12,
+                                 Bitstr_enc* streamdata,
+                                 IsacSaveEncoderData* encData) {
   int k, j;
   int16_t index[PITCH_SUBFRAMES];
   int32_t meangainQ12, CQ17;
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 efed767..741646f 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
@@ -53,12 +53,12 @@
                             int16_t *model,
                             int32_t *sizeQ11,
                             Bitstr_enc *streamdata,
-                            ISAC_SaveEncData_t* encData,
+                            IsacSaveEncoderData* encData,
                             transcode_obj *transcodeParam);
 
 int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17,
                                  Bitstr_enc *streamdata,
-                                 ISAC_SaveEncData_t* encData);
+                                 IsacSaveEncoderData* encData);
 /* decode & dequantize RC */
 int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata,
                                int16_t *RCQ15);
@@ -77,12 +77,12 @@
 
 int WebRtcIsacfix_EncodePitchGain(int16_t *PitchGains_Q12,
                                   Bitstr_enc *streamdata,
-                                  ISAC_SaveEncData_t* encData);
+                                  IsacSaveEncoderData* encData);
 
 int WebRtcIsacfix_EncodePitchLag(int16_t *PitchLagQ7,
                                  int16_t *PitchGain_Q12,
                                  Bitstr_enc *streamdata,
-                                 ISAC_SaveEncData_t* encData);
+                                 IsacSaveEncoderData* encData);
 
 int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
                                   int16_t *PitchGain_Q12);
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 5e8fd4f..922e029 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -135,7 +135,7 @@
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
 
   /* Allocate memory for storing encoder data */
-  ISAC_inst->ISACenc_obj.SaveEnc_ptr = malloc(1 * sizeof(ISAC_SaveEncData_t));
+  ISAC_inst->ISACenc_obj.SaveEnc_ptr = malloc(1 * sizeof(IsacSaveEncoderData));
 
   if (ISAC_inst->ISACenc_obj.SaveEnc_ptr!=NULL) {
     return(0);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
index 0e84cd0..5f3e0c3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
@@ -299,7 +299,7 @@
   /* Used in adaptive mode only */
   int     minBytes;
 
-} ISAC_SaveEncData_t;
+} IsacSaveEncoderData;
 
 typedef struct {
 
@@ -327,14 +327,14 @@
   PostFiltBankstr     interpolatorstr_obj;
 #endif
 
-  ISAC_SaveEncData_t *SaveEnc_ptr;
+  IsacSaveEncoderData *SaveEnc_ptr;
   int16_t         payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
   int16_t         payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
   int16_t         maxPayloadBytes;     /* Maximum allowed number of bits for both 30 and 60 msec packet */
   int16_t         maxRateInBytes;      /* Maximum allowed rate in bytes per 30 msec packet */
   int16_t         enforceFrameSize;    /* If set iSAC will never change packet size */
 
-} ISACFIX_EncInst_t;
+} IsacFixEncoderInstance;
 
 
 typedef struct {
@@ -349,14 +349,14 @@
   PreFiltBankstr      decimatorstr_obj;
 #endif
 
-} ISACFIX_DecInst_t;
+} IsacFixDecoderInstance;
 
 
 
 typedef struct {
 
-  ISACFIX_EncInst_t ISACenc_obj;
-  ISACFIX_DecInst_t ISACdec_obj;
+  IsacFixEncoderInstance ISACenc_obj;
+  IsacFixDecoderInstance ISACdec_obj;
   BwEstimatorstr     bwestimator_obj;
   int16_t         CodingMode;       /* 0 = adaptive; 1 = instantaneous */
   int16_t   errorcode;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
index 845f357..b7319b9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
@@ -42,7 +42,7 @@
                         int16_t codingMode, int16_t
                         bottleneckIndex);
 
-int WebRtcIsac_EncodeStoredDataLb(const ISAC_SaveEncData_t* ISACSavedEnc_obj,
+int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
                                   Bitstr* ISACBitStr_obj, int BWnumber,
                                   float scale);
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
index 5d8fa38..a089f72 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
@@ -1004,7 +1004,7 @@
    The data needed is taken from the structure, where it was stored
    when calling the encoder. */
 
-int WebRtcIsac_EncodeStoredDataLb(const ISAC_SaveEncData_t* ISACSavedEnc_obj,
+int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
                                   Bitstr* ISACBitStr_obj, int BWnumber,
                                   float scale) {
   int ii;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
index 6cd3310..c8d37e5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
@@ -995,7 +995,7 @@
 
 /* Encode LPC in LAR domain. */
 void WebRtcIsac_EncodeLar(double* LPCCoef, Bitstr* streamdata,
-                          ISAC_SaveEncData_t* encData) {
+                          IsacSaveEncoderData* encData) {
   int j, k, n, pos, pos2, poss, offss, offs2;
   int index_s[KLT_ORDER_SHAPE];
   int index_ovr_s[KLT_ORDER_SHAPE];
@@ -1155,7 +1155,7 @@
 
 
 void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi,
-                            Bitstr* streamdata, ISAC_SaveEncData_t* encData) {
+                            Bitstr* streamdata, IsacSaveEncoderData* encData) {
   double lars[KLT_ORDER_GAIN + KLT_ORDER_SHAPE];
   int k;
 
@@ -1233,7 +1233,7 @@
 
 void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
                                 Bitstr* streamdata,
-                                ISAC_SaveEncData_t* encData) {
+                                IsacSaveEncoderData* encData) {
   int j, k, n, pos, pos2, posg, offsg, offs2;
   int index_g[KLT_ORDER_GAIN];
   int index_ovr_g[KLT_ORDER_GAIN];
@@ -1533,7 +1533,7 @@
 /* Quantize & code Pitch Gains. */
 void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12,
                                 Bitstr* streamdata,
-                                ISAC_SaveEncData_t* encData) {
+                                IsacSaveEncoderData* encData) {
   int k, j;
   double C;
   double S[PITCH_SUBFRAMES];
@@ -1678,7 +1678,7 @@
 /* Quantize & code pitch lags. */
 void WebRtcIsac_EncodePitchLag(double* PitchLags, int16_t* PitchGain_Q12,
                                Bitstr* streamdata,
-                               ISAC_SaveEncData_t* encData) {
+                               IsacSaveEncoderData* encData) {
   int k, j;
   double StepSize;
   double C;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h b/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
index 1a86ce2..d715d86 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
@@ -87,11 +87,11 @@
 
 /* quantize & code LPC Coef */
 void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi,
-                            Bitstr* streamdata, ISAC_SaveEncData_t* encData);
+                            Bitstr* streamdata, IsacSaveEncoderData* encData);
 
 void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
                                 Bitstr* streamdata,
-                                ISAC_SaveEncData_t* encData);
+                                IsacSaveEncoderData* encData);
 
 /******************************************************************************
  * WebRtcIsac_EncodeLpcUB()
@@ -177,10 +177,12 @@
 
 void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12,
                                 Bitstr* streamdata,
-                                ISAC_SaveEncData_t* encData);
+                                IsacSaveEncoderData* encData);
 
-void WebRtcIsac_EncodePitchLag(double* PitchLags, int16_t* PitchGain_Q12,
-                               Bitstr* streamdata, ISAC_SaveEncData_t* encData);
+void WebRtcIsac_EncodePitchLag(double* PitchLags,
+                               int16_t* PitchGain_Q12,
+                               Bitstr* streamdata,
+                               IsacSaveEncoderData* encData);
 
 int WebRtcIsac_DecodePitchGain(Bitstr* streamdata,
                                int16_t* PitchGain_Q12);
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
index 62c890c..eb85cf3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -290,7 +290,7 @@
   /* Used in adaptive mode only */
   int         minBytes;
 
-} ISAC_SaveEncData_t;
+} IsacSaveEncoderData;
 
 
 typedef struct {
@@ -315,7 +315,7 @@
   PitchFiltstr        pitchfiltstr_obj;
   PitchAnalysisStruct pitchanalysisstr_obj;
   FFTstr              fftstr_obj;
-  ISAC_SaveEncData_t  SaveEnc_obj;
+  IsacSaveEncoderData SaveEnc_obj;
 
   int                 buffer_index;
   int16_t         current_framesamples;