Fixing DTX unittest for AudioEncoderOpus.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_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 158 matching lines @@
   // Trigger a reset again.
   states.encoder->Reset();
   // Verify that the mode is still kVoip.
   EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
 TEST(AudioEncoderOpusTest, ToggleDtx) {
   auto states = CreateCodec(2);
   // Enable DTX
   EXPECT_TRUE(states.encoder->SetDtx(true));
-  // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
+  EXPECT_TRUE(states.encoder->GetDtx());
   // Turn off DTX.
   EXPECT_TRUE(states.encoder->SetDtx(false));
+  EXPECT_FALSE(states.encoder->GetDtx());
 }
 
 TEST(AudioEncoderOpusTest,
      OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
   auto states = CreateCodec(1);
   // Constants are replicated from audio_states.encoderopus.cc.
   const int kMinBitrateBps = 6000;
   const int kMaxBitrateBps = 512000;
   // Set a too low bitrate.
   states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1,
@@ skipping 347 matching lines @@
 
     // Should encode now.
     states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
                            &encoded);
     EXPECT_GT(encoded.size(), 0u);
     encoded.Clear();
   }
 }
 
 }  // namespace webrtc