Reland "Upconvert various types to int.", neteq portion.

webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 145 matching lines @@
   return true;
 }
 
 int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   DCHECK_EQ(sample_rate_hz, 8000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
+                                 static_cast<int16_t>(encoded_len), decoded,
+                                 &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
 int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
   return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
 int AudioDecoderIlbc::Init() {
   return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
@@ skipping 144 matching lines @@
   WebRtcOpus_DecoderFree(dec_state_);
 }
 
 int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
-                                  static_cast<int16_t>(encoded_len), decoded,
-                                  &temp_type);
+  int ret = WebRtcOpus_Decode(dec_state_, encoded,
+                              static_cast<int16_t>(encoded_len), decoded,
+                              &temp_type);
   if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
 int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
                                               size_t encoded_len,
                                               int sample_rate_hz,
                                               int16_t* decoded,
                                               SpeechType* speech_type) {
   if (!PacketHasFec(encoded, encoded_len)) {
     // This packet is a RED packet.
     return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
                           speech_type);
   }
 
   DCHECK_EQ(sample_rate_hz, 48000);
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
+                                 static_cast<int16_t>(encoded_len), decoded,
+                                 &temp_type);
   if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
 
 int AudioDecoderOpus::Init() {
   return WebRtcOpus_DecoderInit(dec_state_);
 }
 
 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
@@ skipping 242 matching lines @@
     case kDecoderRED:
     case kDecoderAVT:
     case kDecoderArbitrary:
     default: {
       return NULL;
     }
   }
 }
 
 }  // namespace webrtc