| 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,
|
|
|