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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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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);

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