Revert "Upconvert various types to int."

webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h"
 #include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
 
 namespace webrtc {
 
 struct IsacFix {
   typedef ISACFIX_MainStruct instance_type;
   static const bool has_swb = false;
   static const uint16_t kFixSampleRate = 16000;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
-                                int framesize) {
+                                int16_t framesize) {
     return WebRtcIsacfix_Control(inst, rate, framesize);
   }
   static inline int16_t ControlBwe(instance_type* inst,
                                    int32_t rate_bps,
-                                   int frame_size_ms,
+                                   int16_t frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
                                     enforce_frame_size);
   }
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsacfix_Create(inst);
   }
-  static inline int DecodeInternal(instance_type* inst,
-                                   const uint8_t* encoded,
-                                   int16_t len,
-                                   int16_t* decoded,
-                                   int16_t* speech_type) {
+  static inline int16_t DecodeInternal(instance_type* inst,
+                                       const uint8_t* encoded,
+                                       int16_t len,
+                                       int16_t* decoded,
+                                       int16_t* speech_type) {
     return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
   }
   static inline int16_t DecodePlc(instance_type* inst,
                                   int16_t* decoded,
                                   int16_t num_lost_frames) {
     return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
   }
   static inline int16_t DecoderInit(instance_type* inst) {
     return WebRtcIsacfix_DecoderInit(inst);
   }
-  static inline int Encode(instance_type* inst,
-                           const int16_t* speech_in,
-                           uint8_t* encoded) {
+  static inline int16_t Encode(instance_type* inst,
+                               const int16_t* speech_in,
+                               uint8_t* encoded) {
     return WebRtcIsacfix_Encode(inst, speech_in, encoded);
   }
   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
     return WebRtcIsacfix_EncoderInit(inst, coding_mode);
   }
   static inline uint16_t EncSampRate(instance_type* inst) {
     return kFixSampleRate;
   }
 
   static inline int16_t Free(instance_type* inst) {
@@ skipping 72 matching lines @@
   int ErrorCode() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   int PacketDurationRedundant(const uint8_t* encoded,
                               size_t encoded_len) const override;
   bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_