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 27998923f03b6c3a04004e6f946af07d40e7ee6c..c390a90037805b6232eaf8cb0ad5a4a95689457b 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) { |
@@ -145,7 +145,7 @@ class AudioEncoderDecoderMutableIsacFloat |
int16_t* decoded, |
SpeechType* speech_type) override; |
bool HasDecodePlc() const override; |
- int DecodePlc(int num_frames, int16_t* decoded) override; |
+ size_t DecodePlc(size_t num_frames, int16_t* decoded) override; |
int Init() override; |
int IncomingPacket(const uint8_t* payload, |
size_t payload_len, |