Make FakeDecodeFromFile handle codec-internal CNG

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h

 /*
  *  Copyright (c) 2016 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 32 matching lines @@
   int SampleRateHz() const override { return sample_rate_hz_; }
 
   size_t Channels() const override { return stereo_ ? 2 : 1; }
 
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
                      int16_t* decoded,
                      SpeechType* speech_type) override;
 
-  // Helper method. Writes |timestamp| and |samples| to |encoded| in a format
-  // that the FakeDecpdeFromFile decoder will understand. |encoded| must be at
-  // least 8 bytes long.
+  // Helper method. Writes |timestamp|, |samples| and
+  // |original_payload_size_bytes| to |encoded| in a format that the
+  // FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
+  // bytes long.
   static void PrepareEncoded(uint32_t timestamp,
                              size_t samples,
+                             size_t original_payload_size_bytes,
                              rtc::ArrayView<uint8_t> encoded);
 
  private:
   std::unique_ptr<InputAudioFile> input_;
   rtc::Optional<uint32_t> next_timestamp_from_input_;
   const int sample_rate_hz_;
   const bool stereo_;
+  size_t last_decoded_length_ = 0;
+  bool cng_mode_ = false;
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_