Regression test for issue where Opus DTX status was being forgotten.

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  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 109 matching lines @@
   // true when asked to disable FEC and false when asked to enable it (meaning
   // that FEC isn't supported).
   virtual bool SetFec(bool enable);
 
   // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
   // able to comply. The default implementation returns true when asked to
   // disable DTX and false when asked to enable it (meaning that DTX isn't
   // supported).
   virtual bool SetDtx(bool enable);
 
+  // Returns the status of codec-internal DTX. The default implementation always
+  // returns false.
+  virtual bool GetDtx();
+
   // Sets the application mode. Returns true if the codec was able to comply.
   // The default implementation just returns false.
   enum class Application { kSpeech, kAudio };
   virtual bool SetApplication(Application application);
 
   // Tells the encoder about the highest sample rate the decoder is expected to
   // use when decoding the bitstream. The encoder would typically use this
   // information to adjust the quality of the encoding. The default
   // implementation does nothing.
   virtual void SetMaxPlaybackRate(int frequency_hz);
@@ skipping 20 matching lines @@
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_