Match existing type usage better.

webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc

 /*
  *  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.
  */
 
@@ skipping 178 matching lines @@
     // In this test, we let the target packet loss rate match the actual rate.
     actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
     // Run every mode a certain time.
     time_now_ms = 0;
     fec_frames = 0;
     while (time_now_ms < kDurationMs) {
       // Encode & decode.
       EncodeABlock();
 
       // Check if payload has FEC.
-      int16_t fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
 
       // If FEC is disabled or the target packet loss rate is set to 0, there
       // should be no FEC in the bit stream.
       if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) {
         EXPECT_EQ(fec, 0);
       } else if (fec == 1) {
         fec_frames++;
       }
 
       lost_previous = lost_current;
@@ skipping 20 matching lines @@
      ::std::tr1::make_tuple(1, 32000, string("audio_coding/testfile32kHz"),
                             string("pcm")),
      ::std::tr1::make_tuple(2, 64000, string("audio_coding/teststereo32kHz"),
                             string("pcm"))};
 
 // 64 kbps, stereo
 INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc