Index: webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h |
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h |
index c0f3b11a09ca82e7dc5127f214bfb78af112af8f..1bfd149b27b4a0cb9faaf4ff29e23a9cf456d6d1 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h |
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h |
@@ -39,14 +39,14 @@ struct IsacFloat { |
} |
static inline int DecodeInternal(instance_type* inst, |
const uint8_t* encoded, |
- int16_t len, |
+ size_t len, |
int16_t* decoded, |
int16_t* speech_type) { |
return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type); |
} |
- static inline int16_t DecodePlc(instance_type* inst, |
- int16_t* decoded, |
- int16_t num_lost_frames) { |
+ static inline size_t DecodePlc(instance_type* inst, |
+ int16_t* decoded, |
+ size_t num_lost_frames) { |
return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames); |
} |
@@ -102,7 +102,7 @@ struct IsacFloat { |
} |
static inline int16_t UpdateBwEstimate(instance_type* 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) { |