Reland Remove the deprecated EncodeInternal interface from AudioEncoder

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 36 matching lines @@
     static const int kDefaultComplexity = 5;
 #else
     static const int kDefaultComplexity = 9;
 #endif
   };
 
   explicit AudioEncoderOpus(const Config& config);
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   ~AudioEncoderOpus() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
 
   void Reset() override;
   bool SetFec(bool enable) override;
 
   // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
   // being inactive. During that, it still sends 2 packets (one for content, one
   // for signaling) about every 400 ms.
   bool SetDtx(bool enable) override;
 
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
 
   // Getters for testing.
   double packet_loss_rate() const { return packet_loss_rate_; }
   ApplicationMode application() const { return config_.application; }
   bool dtx_enabled() const { return config_.dtx_enabled; }
 
-protected:
+ protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
 
  private:
   size_t Num10msFramesPerPacket() const;
   size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
   bool RecreateEncoderInstance(const Config& config);
 
   Config config_;
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_