AudioDecoder: Replace Init() with Reset()

webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_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 30 matching lines @@
 
 void IsacSpeedTest::SetUp() {
   AudioCodecSpeedTest::SetUp();
 
   // Check whether the allocated buffer for the bit stream is large enough.
   EXPECT_GE(max_bytes_, static_cast<size_t>(STREAM_MAXW16_60MS));
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
   EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
-  EXPECT_EQ(0, WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_));
+  WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_);
   // Set bitrate and block length.
   EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
                                      block_duration_ms_));
 }
 
 void IsacSpeedTest::TearDown() {
   AudioCodecSpeedTest::TearDown();
   // Free memory.
   EXPECT_EQ(0, WebRtcIsacfix_Free(ISACFIX_main_inst_));
 }
@@ skipping 40 matching lines @@
 }
 
 const coding_param param_set[] =
     {::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
                             string("pcm"), true)};
 
 INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc