Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int>

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_
 #define WEBRTC_CALL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 
 const char* Version();
 
@@ skipping 40 matching lines @@
 class Call {
  public:
   struct Config {
     static const int kDefaultStartBitrateBps;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
-      int max_bitrate_bps = -1;
+      rtc::Optional<int> max_bitrate_bps;
     } bitrate_config;
 
     // 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;
   };
@@ skipping 50 matching lines @@
                                          NetworkState state) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_