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authorpbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-04-09 00:28:06 +0000
committerpbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-04-09 00:28:06 +0000
commit0946a56023d821e0deca04029bb016ae1f23aa82 (patch)
tree504a7ac85144a61e871e5858ffdae54fddd99815 /webrtc/modules/audio_coding/codecs/g722
parent6faf71d27b89ce4fc29ebd8148fc2ffd90983a61 (diff)
downloadwebrtc-0946a56023d821e0deca04029bb016ae1f23aa82.tar.gz
WebRtc_Word32 => int32_t etc. in audio_coding/
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1271006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'webrtc/modules/audio_coding/codecs/g722')
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/g722_decode.c16
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h8
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/g722_encode.c18
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/g722_interface.c34
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h32
-rw-r--r--webrtc/modules/audio_coding/codecs/g722/test/testG722.cc16
6 files changed, 62 insertions, 62 deletions
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
index 499cc8fa30..e62af981fd 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
@@ -49,12 +49,12 @@
#define TRUE (!FALSE)
#endif
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+static __inline int16_t saturate(int32_t amp)
{
- WebRtc_Word16 amp16;
+ int16_t amp16;
/* Hopefully this is optimised for the common case - not clipping */
- amp16 = (WebRtc_Word16) amp;
+ amp16 = (int16_t) amp;
if (amp == amp16)
return amp16;
if (amp > WEBRTC_INT16_MAX)
@@ -190,8 +190,8 @@ int WebRtc_g722_decode_release(g722_decode_state_t *s)
}
/*- End of function --------------------------------------------------------*/
-int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
- const WebRtc_UWord8 g722_data[], int len)
+int WebRtc_g722_decode(g722_decode_state_t *s, int16_t amp[],
+ const uint8_t g722_data[], int len)
{
static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
@@ -372,14 +372,14 @@ int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
if (s->itu_test_mode)
{
- amp[outlen++] = (WebRtc_Word16) (rlow << 1);
- amp[outlen++] = (WebRtc_Word16) (rhigh << 1);
+ amp[outlen++] = (int16_t) (rlow << 1);
+ amp[outlen++] = (int16_t) (rhigh << 1);
}
else
{
if (s->eight_k)
{
- amp[outlen++] = (WebRtc_Word16) (rlow << 1);
+ amp[outlen++] = (int16_t) (rlow << 1);
}
else
{
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 d2d19b04b1..ef279ac5a8 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -138,8 +138,8 @@ g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
int options);
int WebRtc_g722_encode_release(g722_encode_state_t *s);
int WebRtc_g722_encode(g722_encode_state_t *s,
- WebRtc_UWord8 g722_data[],
- const WebRtc_Word16 amp[],
+ uint8_t g722_data[],
+ const int16_t amp[],
int len);
g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
@@ -147,8 +147,8 @@ g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
int options);
int WebRtc_g722_decode_release(g722_decode_state_t *s);
int WebRtc_g722_decode(g722_decode_state_t *s,
- WebRtc_Word16 amp[],
- const WebRtc_UWord8 g722_data[],
+ int16_t amp[],
+ const uint8_t g722_data[],
int len);
#ifdef __cplusplus
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_encode.c b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
index 7487b64c7f..5b07615a0c 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
@@ -48,12 +48,12 @@
#define TRUE (!FALSE)
#endif
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+static __inline int16_t saturate(int32_t amp)
{
- WebRtc_Word16 amp16;
+ int16_t amp16;
/* Hopefully this is optimised for the common case - not clipping */
- amp16 = (WebRtc_Word16) amp;
+ amp16 = (int16_t) amp;
if (amp == amp16)
return amp16;
if (amp > WEBRTC_INT16_MAX)
@@ -191,10 +191,10 @@ int WebRtc_g722_encode_release(g722_encode_state_t *s)
*/
//#define RUN_LIKE_REFERENCE_G722
#ifdef RUN_LIKE_REFERENCE_G722
-WebRtc_Word16 limitValues (WebRtc_Word16 rl)
+int16_t limitValues (int16_t rl)
{
- WebRtc_Word16 yl;
+ int16_t yl;
yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
@@ -202,8 +202,8 @@ WebRtc_Word16 limitValues (WebRtc_Word16 rl)
}
#endif
-int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
- const WebRtc_Word16 amp[], int len)
+int WebRtc_g722_encode(g722_encode_state_t *s, uint8_t g722_data[],
+ const int16_t amp[], int len)
{
static const int q6[32] =
{
@@ -418,14 +418,14 @@ int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
s->out_bits += s->bits_per_sample;
if (s->out_bits >= 8)
{
- g722_data[g722_bytes++] = (WebRtc_UWord8) (s->out_buffer & 0xFF);
+ g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
s->out_bits -= 8;
s->out_buffer >>= 8;
}
}
else
{
- g722_data[g722_bytes++] = (WebRtc_UWord8) code;
+ g722_data[g722_bytes++] = (uint8_t) code;
}
}
return g722_bytes;
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index d559014225..7075669f59 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -17,7 +17,7 @@
#include "typedefs.h"
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
{
*G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
if (*G722enc_inst!=NULL) {
@@ -27,7 +27,7 @@ WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
}
}
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
{
// Create and/or reset the G.722 encoder
// Bitrate 64 kbps and wideband mode (2)
@@ -40,16 +40,16 @@ WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
}
}
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
// Free encoder memory
return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
}
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded)
+int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+ int16_t *speechIn,
+ int16_t len,
+ int16_t *encoded)
{
unsigned char *codechar = (unsigned char*) encoded;
// Encode the input speech vector
@@ -57,7 +57,7 @@ WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
codechar, speechIn, len);
}
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
{
*G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
if (*G722dec_inst!=NULL) {
@@ -67,7 +67,7 @@ WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
}
}
-WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
+int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
{
// Create and/or reset the G.722 decoder
// Bitrate 64 kbps and wideband mode (2)
@@ -80,25 +80,25 @@ WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
}
}
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
// Free encoder memory
return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
}
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
+int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ int16_t *encoded,
+ int16_t len,
+ int16_t *decoded,
+ int16_t *speechType)
{
// Decode the G.722 encoder stream
*speechType=G722_WEBRTC_SPEECH;
return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
- decoded, (WebRtc_UWord8*) encoded, len);
+ decoded, (uint8_t*) encoded, len);
}
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len)
+int16_t WebRtcG722_Version(char *versionStr, short len)
{
// Get version string
char version[30] = "2.0.0\n";
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 e50d66f56c..0948a1831c 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -43,7 +43,7 @@ extern "C" {
* Return value : 0 - Ok
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
/****************************************************************************
@@ -59,7 +59,7 @@ WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
/****************************************************************************
@@ -73,7 +73,7 @@ WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
* Return value : 0 - Ok
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
@@ -95,10 +95,10 @@ WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded);
+int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+ int16_t *speechIn,
+ int16_t len,
+ int16_t *encoded);
/****************************************************************************
@@ -112,7 +112,7 @@ WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
* Return value : 0 - Ok
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
/****************************************************************************
@@ -128,7 +128,7 @@ WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
+int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
/****************************************************************************
@@ -143,7 +143,7 @@ WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
/****************************************************************************
@@ -167,11 +167,11 @@ WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
* -1 - Error
*/
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
+int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ int16_t *encoded,
+ int16_t len,
+ int16_t *decoded,
+ int16_t *speechType);
/****************************************************************************
* WebRtcG722_Version(...)
@@ -179,7 +179,7 @@ WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
* Get a string with the current version of the codec
*/
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len);
+int16_t WebRtcG722_Version(char *versionStr, short len);
#ifdef __cplusplus
diff --git a/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
index d2fdca3a85..d51301d511 100644
--- a/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -29,11 +29,11 @@ typedef struct WebRtcG722EncInst G722EncInst;
typedef struct WebRtcG722DecInst G722DecInst;
/* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length)
+int readframe(int16_t *data, FILE *inp, int length)
{
short k, rlen, status = 0;
- rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
+ rlen = (short)fread(data, sizeof(int16_t), length, inp);
if (rlen < length) {
for (k = rlen; k < length; k++)
data[k] = 0;
@@ -49,7 +49,7 @@ int main(int argc, char* argv[])
FILE *inp, *outbitp, *outp;
int framecnt, endfile;
- WebRtc_Word16 framelength = 160;
+ int16_t framelength = 160;
G722EncInst *G722enc_inst;
G722DecInst *G722dec_inst;
int err;
@@ -59,11 +59,11 @@ int main(int argc, char* argv[])
double runtime = 0;
double length_file;
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 shortdata[960];
- WebRtc_Word16 decoded[960];
- WebRtc_Word16 streamdata[80*3];
- WebRtc_Word16 speechType[1];
+ int16_t stream_len = 0;
+ int16_t shortdata[960];
+ int16_t decoded[960];
+ int16_t streamdata[80*3];
+ int16_t speechType[1];
/* handling wrong input arguments in the command line */
if (argc!=5) {