Fix bug where min transmit bitrate wasn't working

webrtc/call.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_CALL_H_
@@ skipping 73 matching lines @@
     // AudioState which is possibly shared between multiple calls.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     rtc::scoped_refptr<AudioState> audio_state;
 
     // Audio Processing Module to be used in this call.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     AudioProcessing* audio_processing = nullptr;
   };
 
   struct Stats {
-    int send_bandwidth_bps = 0;
-    int recv_bandwidth_bps = 0;
+    int send_bandwidth_bps = 0;        // Estimated available send bandwidth.
+    int min_transmit_bitrate_bps = 0;  // Bitrate to send, padding as needed.
+    int recv_bandwidth_bps = 0;        // Estimated available receive bandwidth.
     int64_t pacer_delay_ms = 0;
     int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
 
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
 
@@ skipping 40 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_