Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/codecs/opus/audio_encoder_mutable_opus_test.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.
  */
 
+// TODO(kwiberg): Merge these tests into audio_encoder_opus_unittest.cc
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 
 namespace webrtc {
 namespace acm2 {
 
 #ifdef WEBRTC_CODEC_OPUS
 namespace {
 const CodecInst kDefaultOpusCodecInst = {105, "opus", 48000, 960, 1, 32000};
 }  // namespace
 
 class AudioEncoderMutableOpusTest : public ::testing::Test {
  protected:
   AudioEncoderMutableOpusTest() : codec_inst_(kDefaultOpusCodecInst) {}
 
   void CreateCodec(int num_channels) {
     codec_inst_.channels = num_channels;
-    encoder_.reset(new AudioEncoderMutableOpus(codec_inst_));
+    encoder_.reset(new AudioEncoderOpus(codec_inst_));
     auto expected_app =
         num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
     EXPECT_EQ(expected_app, encoder_->application());
   }
 
   CodecInst codec_inst_;
-  rtc::scoped_ptr<AudioEncoderMutableOpus> encoder_;
+  rtc::scoped_ptr<AudioEncoderOpus> encoder_;
 };
 
 TEST_F(AudioEncoderMutableOpusTest, DefaultApplicationModeMono) {
   CreateCodec(1);
 }
 
 TEST_F(AudioEncoderMutableOpusTest, DefaultApplicationModeStereo) {
   CreateCodec(2);
 }
 
 TEST_F(AudioEncoderMutableOpusTest, ChangeApplicationMode) {
   CreateCodec(2);
-  EXPECT_TRUE(
-      encoder_->SetApplication(AudioEncoderMutable::kApplicationSpeech));
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
   EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
 }
 
 TEST_F(AudioEncoderMutableOpusTest, ResetWontChangeApplicationMode) {
   CreateCodec(2);
 
   // Trigger a reset.
   encoder_->Reset();
   // Verify that the mode is still kAudio.
   EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
 
   // Now change to kVoip.
-  EXPECT_TRUE(
-      encoder_->SetApplication(AudioEncoderMutable::kApplicationSpeech));
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
   EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
 
   // Trigger a reset again.
   encoder_->Reset();
   // Verify that the mode is still kVoip.
   EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
 }
 
 TEST_F(AudioEncoderMutableOpusTest, ToggleDtx) {
   CreateCodec(2);
@@ skipping 25 matching lines @@
   // Set rates from 1000 up to 32000 bps.
   for (int rate = 1000; rate <= 32000; rate += 1000) {
     encoder_->SetTargetBitrate(rate);
     EXPECT_EQ(rate, encoder_->GetTargetBitrate());
   }
 }
 #endif  // WEBRTC_CODEC_OPUS
 
 }  // namespace acm2
 }  // namespace webrtc