AudioDecoder: Replace Init() with Reset()

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.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 426 matching lines @@
                size_t /* max_decoded_bytes */,
                int16_t* decoded,
                SpeechType* speech_type) override {
       for (size_t i = 0; i < encoded_len; ++i) {
         decoded[i] = next_value_++;
       }
       *speech_type = kSpeech;
       return encoded_len;
     }
 
-    virtual int Init() {
-      next_value_ = 1;
-      return 0;
-    }
+    void Reset() override { next_value_ = 1; }
 
     size_t Channels() const override { return 1; }
 
     uint16_t next_value() const { return next_value_; }
 
    private:
     int16_t next_value_;
   } decoder_;
 
   EXPECT_EQ(NetEq::kOK,
@@ skipping 56 matching lines @@
   const size_t kPayloadLengthBytes = kPayloadLengthSamples;
   uint8_t payload[kPayloadLengthBytes] = {0};
   WebRtcRTPHeader rtp_header;
   rtp_header.header.payloadType = kPayloadType;
   rtp_header.header.sequenceNumber = 0x1234;
   rtp_header.header.timestamp = 0x12345678;
   rtp_header.header.ssrc = 0x87654321;
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
   int16_t dummy_output[kPayloadLengthSamples] = {0};
   // The below expectation will make the mock decoder write
   // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
   EXPECT_CALL(mock_decoder,
               Decode(Pointee(0), kPayloadLengthBytes, kSampleRateHz, _, _, _))
       .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
@@ skipping 145 matching lines @@
   int16_t dummy_output[kPayloadLengthSamples] = {0};
 
   WebRtcRTPHeader rtp_header;
   rtp_header.header.payloadType = kPayloadType;
   rtp_header.header.sequenceNumber = 0x1234;
   rtp_header.header.timestamp = 0x12345678;
   rtp_header.header.ssrc = 0x87654321;
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
 
   // Pointee(x) verifies that first byte of the payload equals x, this makes it
   // possible to verify that the correct payload is fed to Decode().
   EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes,
                                    kSampleRateKhz * 1000, _, _, _))
       .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
@@ skipping 118 matching lines @@
   uint8_t payload[kPayloadLengthBytes]= {0};
   int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
   WebRtcRTPHeader rtp_header;
   rtp_header.header.payloadType = kPayloadType;
   rtp_header.header.sequenceNumber = 0x1234;
   rtp_header.header.timestamp = 0x12345678;
   rtp_header.header.ssrc = 0x87654321;
 
   class MockAudioDecoder : public AudioDecoder {
    public:
-    int Init() override {
-      return 0;
-    }
+    void Reset() override {}
     MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
     MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
                                      SpeechType*));
     size_t Channels() const override { return kChannels; }
   } decoder_;
 
   const uint8_t kFirstPayloadValue = 1;
   const uint8_t kSecondPayloadValue = 2;
 
   EXPECT_CALL(decoder_, PacketDuration(Pointee(kFirstPayloadValue),
@@ skipping 56 matching lines @@
   EXPECT_EQ(kChannels, num_channels);
 
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(kMaxOutputSize, output,
                                          &samples_per_channel, &num_channels,
                                          &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
   EXPECT_EQ(kChannels, num_channels);
 }
 
 }  // namespace webrtc