Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.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 41 matching lines @@
 
   explicit AudioEncoderDecoderIsacT(const Config& config);
   ~AudioEncoderDecoderIsacT() override;
 
   // AudioEncoder public methods.
   int SampleRateHz() const override;
   int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
   int Num10MsFramesInNextPacket() const override;
   int Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
 
   // AudioDecoder methods.
   bool HasDecodePlc() const override;
   int DecodePlc(int num_frames, int16_t* decoded) override;
   int Init() override;
   int IncomingPacket(const uint8_t* payload,
                      size_t payload_len,
                      uint16_t rtp_sequence_number,
                      uint32_t rtp_timestamp,
                      uint32_t arrival_timestamp) override;
@@ skipping 34 matching lines @@
 
   // Have we accepted input but not yet emitted it in a packet?
   bool packet_in_progress_ GUARDED_BY(lock_);
 
   // Timestamp of the first input of the currently in-progress packet.
   uint32_t packet_timestamp_ GUARDED_BY(lock_);
 
   // Timestamp of the previously encoded packet.
   uint32_t last_encoded_timestamp_ GUARDED_BY(lock_);
 
+  const int target_bitrate_bps_;
+
   DISALLOW_COPY_AND_ASSIGN(AudioEncoderDecoderIsacT);
 };
 
 struct CodecInst;
 
 class AudioEncoderDecoderMutableIsac : public AudioEncoderMutable,
                                        public AudioDecoder {
  public:
   virtual void UpdateSettings(const CodecInst& codec_inst) = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_