| Index: webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
|
| 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 429fc6b6bf833cdadaf6fd53e3aab0e262471012..0597de8ae81e6464db9c4f02acf6edbf088df8af 100644
|
| --- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
|
| +++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
|
| @@ -11,6 +11,8 @@
|
| #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
|
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
|
|
|
| +#include <stddef.h>
|
| +
|
| #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
|
| #include "webrtc/typedefs.h"
|
|
|
| @@ -186,7 +188,7 @@ extern "C" {
|
| int16_t WebRtcIsac_UpdateBwEstimate(
|
| ISACStruct* ISAC_main_inst,
|
| const uint8_t* encoded,
|
| - int32_t packet_size,
|
| + size_t packet_size,
|
| uint16_t rtp_seq_number,
|
| uint32_t send_ts,
|
| uint32_t arr_ts);
|
| @@ -215,7 +217,7 @@ extern "C" {
|
| int WebRtcIsac_Decode(
|
| ISACStruct* ISAC_main_inst,
|
| const uint8_t* encoded,
|
| - int16_t len,
|
| + size_t len,
|
| int16_t* decoded,
|
| int16_t* speechType);
|
|
|
| @@ -235,14 +237,13 @@ extern "C" {
|
| * Output:
|
| * - decoded : The decoded vector.
|
| *
|
| - * Return value : >0 - number of samples in decoded PLC vector
|
| - * -1 - Error
|
| + * Return value : Number of samples in decoded PLC vector
|
| */
|
|
|
| - int16_t WebRtcIsac_DecodePlc(
|
| + size_t WebRtcIsac_DecodePlc(
|
| ISACStruct* ISAC_main_inst,
|
| int16_t* decoded,
|
| - int16_t noOfLostFrames);
|
| + size_t noOfLostFrames);
|
|
|
|
|
| /******************************************************************************
|
| @@ -704,7 +705,7 @@ extern "C" {
|
| int WebRtcIsac_DecodeRcu(
|
| ISACStruct* ISAC_main_inst,
|
| const uint8_t* encoded,
|
| - int16_t len,
|
| + size_t len,
|
| int16_t* decoded,
|
| int16_t* speechType);
|
|
|
|
|