Change frame length on very low bandwidth.

webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.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 14 matching lines @@
  public:
   struct Config {
     struct FrameLengthChange {
       FrameLengthChange(int from_frame_length_ms, int to_frame_length_ms);
       bool operator<(const FrameLengthChange& rhs) const;
       int from_frame_length_ms;
       int to_frame_length_ms;
     };
     Config(const std::vector<int>& encoder_frame_lengths_ms,
            int initial_frame_length_ms,
+           int min_encoder_bitrate_bps,
            float fl_increasing_packet_loss_fraction,
            float fl_decreasing_packet_loss_fraction,
            std::map<FrameLengthChange, int> fl_changing_bandwidths_bps);
     Config(const Config& other);
     ~Config();
     std::vector<int> encoder_frame_lengths_ms;
     int initial_frame_length_ms;
+    int min_encoder_bitrate_bps;
     // Uplink packet loss fraction below which frame length can increase.
     float fl_increasing_packet_loss_fraction;
     // Uplink packet loss fraction below which frame length should decrease.
     float fl_decreasing_packet_loss_fraction;
     std::map<FrameLengthChange, int> fl_changing_bandwidths_bps;
   };
 
   explicit FrameLengthController(const Config& config);
 
   ~FrameLengthController() override;
@@ skipping 10 matching lines @@
       const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const;
 
   const Config config_;
 
   std::vector<int>::const_iterator frame_length_ms_;
 
   rtc::Optional<int> uplink_bandwidth_bps_;
 
   rtc::Optional<float> uplink_packet_loss_fraction_;
 
+  rtc::Optional<size_t> overhead_bytes_per_packet_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(FrameLengthController);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_