Adding cbr support for Opus

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 38 matching lines @@
     // Otherwise, returns the current complexity depending on whether the
     // current bitrate is above or below complexity_threshold_bps.
     rtc::Optional<int> GetNewComplexity() const;
 
     int frame_size_ms = 20;
     size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
     rtc::Optional<int> bitrate_bps;  // Unset means to use default value.
     bool fec_enabled = false;
+    bool cbr_enabled = false;
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
     // This value may change in the struct's constructor.
     int low_rate_complexity = kDefaultComplexity;
     // low_rate_complexity is used when the bitrate is below this threshold.
     int complexity_threshold_bps = 12500;
     int complexity_threshold_window_bps = 1500;
     bool dtx_enabled = false;
     std::vector<int> supported_frame_lengths_ms;
     const Clock* clock = Clock::GetRealTimeClock();
@@ skipping 106 matching lines @@
   rtc::Optional<size_t> overhead_bytes_per_packet_;
   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
   rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_