Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.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 1056 matching lines @@
                          int frame_size_rtp_timestamps) {
     payload_type_ = payload_type;
     frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
     return send_test_->RegisterCodec(payload_name,
                                      sampling_freq_hz,
                                      channels,
                                      payload_type,
                                      frame_size_samples);
   }
 
-  bool RegisterExternalSendCodec(AudioEncoderMutable* external_speech_encoder,
+  bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
                                  int payload_type) {
     payload_type_ = payload_type;
     frame_size_rtp_timestamps_ =
         external_speech_encoder->Num10MsFramesInNextPacket() *
         external_speech_encoder->RtpTimestampRateHz() / 100;
     return send_test_->RegisterExternalCodec(external_speech_encoder);
   }
 
   // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
   // before calling this method.
@@ skipping 81 matching lines @@
                  int codec_frame_size_rtp_timestamps) {
     ASSERT_TRUE(SetUpSender());
     ASSERT_TRUE(RegisterSendCodec(codec_name,
                                   codec_sample_rate_hz,
                                   channels,
                                   payload_type,
                                   codec_frame_size_samples,
                                   codec_frame_size_rtp_timestamps));
   }
 
-  void SetUpTestExternalEncoder(AudioEncoderMutable* external_speech_encoder,
+  void SetUpTestExternalEncoder(AudioEncoder* external_speech_encoder,
                                 int payload_type) {
     ASSERT_TRUE(SetUpSender());
     ASSERT_TRUE(
         RegisterExternalSendCodec(external_speech_encoder, payload_type));
   }
 
   rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
   rtc::scoped_ptr<test::InputAudioFile> audio_source_;
   uint32_t frame_size_rtp_timestamps_;
   int packet_count_;
@@ skipping 404 matching lines @@
 TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
   Run(32000, 64000, 64000);
 }
 
 TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
   CodecInst codec_inst;
   codec_inst.channels = 1;
   codec_inst.pacsize = 160;
   codec_inst.pltype = 0;
-  AudioEncoderMutablePcmU encoder(codec_inst);
-  MockAudioEncoderMutable mock_encoder;
+  AudioEncoderPcmU encoder(codec_inst);
+  MockAudioEncoder mock_encoder;
   // Set expectations on the mock encoder and also delegate the calls to the
   // real encoder.
+  EXPECT_CALL(mock_encoder, MaxEncodedBytes())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::MaxEncodedBytes));
+  EXPECT_CALL(mock_encoder, SampleRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
+  EXPECT_CALL(mock_encoder, NumChannels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
+  EXPECT_CALL(mock_encoder, RtpTimestampRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
   EXPECT_CALL(mock_encoder, Num10MsFramesInNextPacket())
       .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(
-          &encoder, &AudioEncoderMutablePcmU::Num10MsFramesInNextPacket));
+      .WillRepeatedly(
+          Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
   EXPECT_CALL(mock_encoder, Max10MsFramesInAPacket())
       .Times(AtLeast(1))
       .WillRepeatedly(
-          Invoke(&encoder, &AudioEncoderMutablePcmU::Max10MsFramesInAPacket));
-  EXPECT_CALL(mock_encoder, SampleRateHz())
+          Invoke(&encoder, &AudioEncoderPcmU::Max10MsFramesInAPacket));
+  EXPECT_CALL(mock_encoder, GetTargetBitrate())
       .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&encoder, &AudioEncoderMutablePcmU::SampleRateHz));
-  EXPECT_CALL(mock_encoder, NumChannels())
-      .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&encoder, &AudioEncoderMutablePcmU::NumChannels));
-  EXPECT_CALL(mock_encoder, MaxEncodedBytes())
-      .Times(AtLeast(1))
-      .WillRepeatedly(
-          Invoke(&encoder, &AudioEncoderMutablePcmU::MaxEncodedBytes));
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
   EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
       .Times(AtLeast(1))
-      .WillRepeatedly(
-          Invoke(&encoder, &AudioEncoderMutablePcmU::EncodeInternal));
-  EXPECT_CALL(mock_encoder, GetTargetBitrate())
-      .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(
-          &encoder, &AudioEncoderMutablePcmU::GetTargetBitrate));
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));
   ASSERT_NO_FATAL_FAILURE(
       SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
   Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
       50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
 // This test fixture is implemented to run ACM and change the desired output
 // frequency during the call. The input packets are simply PCM16b-wb encoded
 // payloads with a constant value of |kSampleValue|. The test fixture itself
 // acts as PacketSource in between the receive test class and the constant-
@@ skipping 103 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc