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); |