AudioDecoder: Replace Init() with Reset()

webrtc/modules/audio_coding/codecs/isac/unittest.cc

 /*
  *  Copyright (c) 2015 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 93 matching lines @@
 void TestGetSetBandwidthInfo(const int16_t* speech_data,
                              int rate_bits_per_second,
                              int sample_rate_hz,
                              int frame_size_ms) {
   const int bit_rate = 32000;
 
   // Conjoined encoder/decoder pair:
   typename T::instance_type* encdec;
   ASSERT_EQ(0, T::Create(&encdec));
   ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
-  ASSERT_EQ(0, T::DecoderInit(encdec));
+  T::DecoderInit(encdec);
   ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
   if (adaptive)
     ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
   else
     ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms));
 
   // Disjoint encoder/decoder pair:
   typename T::instance_type* enc;
   ASSERT_EQ(0, T::Create(&enc));
   ASSERT_EQ(0, T::EncoderInit(enc, adaptive ? 0 : 1));
   ASSERT_EQ(0, T::SetEncSampRate(enc, sample_rate_hz));
   if (adaptive)
     ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false));
   else
     ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
   typename T::instance_type* dec;
   ASSERT_EQ(0, T::Create(&dec));
-  ASSERT_EQ(0, T::DecoderInit(dec));
+  T::DecoderInit(dec);
   T::SetInitialBweBottleneck(dec, bit_rate);
   T::SetEncSampRateInDecoder(dec, sample_rate_hz);
 
   // 0. Get initial BW info from decoder.
   auto bi = GetBwInfo<T>(dec);
 
   BoundedCapacityChannel channel1(sample_rate_hz, rate_bits_per_second),
       channel2(sample_rate_hz, rate_bits_per_second);
 
   int elapsed_time_ms = 0;
@@ skipping 102 matching lines @@
             for (int frame_size : frame_sizes)
               if (!(sample_rate == 32000 && frame_size == 60))
                 cases.push_back(
                     {type, adaptive, channel_rate, sample_rate, frame_size});
   return cases;
 }
 
 INSTANTIATE_TEST_CASE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
 
 }  // namespace webrtc