Revert "Upconvert various types to int."

webrtc/modules/audio_coding/neteq/neteq_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 1253 matching lines @@
     assert(decoder);  // At this point, we must have a decoder object.
     // The number of channels in the |sync_buffer_| should be the same as the
     // number decoder channels.
     assert(sync_buffer_->Channels() == decoder->Channels());
     assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
     assert(*operation == kNormal || *operation == kAccelerate ||
            *operation == kFastAccelerate || *operation == kMerge ||
            *operation == kPreemptiveExpand);
     packet_list->pop_front();
     size_t payload_length = packet->payload_length;
-    int decode_length;
+    int16_t decode_length;
     if (packet->sync_packet) {
       // Decode to silence with the same frame size as the last decode.
       LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
           " ts=" << packet->header.timestamp <<
           ", sn=" << packet->header.sequenceNumber <<
           ", pt=" << static_cast<int>(packet->header.payloadType) <<
           ", ssrc=" << packet->header.ssrc <<
           ", len=" << packet->payload_length;
       memset(&decoded_buffer_[*decoded_length], 0,
              decoder_frame_length_ * decoder->Channels() *
@@ skipping 700 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc