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

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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 64 matching lines @@
       if (i > 0) {
         ost << ", ";
       }
       ost << vals[i].ToString();
     }
     ost << "]";
     return ost.str();
 }
 
 template <typename T>
-static T MinPositive(T a, T b) {
-  if (a <= 0) {
+static rtc::Optional<T> OptionalMin(rtc::Optional<T> a, rtc::Optional<T> b) {
+  if (!a) {
     return b;
   }
-  if (b <= 0) {
+  if (!b) {
     return a;
   }
-  return std::min(a, b);
+
+  return rtc::Optional<T>(std::min(*a, *b));
 }
 
 // Construction-time settings, passed to
 // MediaControllerInterface::Create, and passed on when creating
 // MediaChannels.
 struct MediaConfig {
   // Set DSCP value on packets. This flag comes from the
   // PeerConnection constraint 'googDscp'.
   bool enable_dscp = false;
 
@@ skipping 742 matching lines @@
 
 // TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
 // encapsulate all the parameters needed for an RtpSender.
 template <class Codec>
 struct RtpSendParameters : RtpParameters<Codec> {
   std::string ToString() const override {
     std::ostringstream ost;
     ost << "{";
     ost << "codecs: " << VectorToString(this->codecs) << ", ";
     ost << "extensions: " << VectorToString(this->extensions) << ", ";
-    ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
+    ost << "max_bitrate_bps: " << max_bitrate_bps << ", ";
     ost << "}";
     return ost.str();
   }
 
-  int max_bandwidth_bps = -1;
+  rtc::Optional<int> max_bitrate_bps;
 };
 
 struct AudioSendParameters : RtpSendParameters<AudioCodec> {
   std::string ToString() const override {
     std::ostringstream ost;
     ost << "{";
     ost << "codecs: " << VectorToString(this->codecs) << ", ";
     ost << "extensions: " << VectorToString(this->extensions) << ", ";
-    ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
+    ost << "max_bitrate_bps: " << max_bitrate_bps << ", ";
     ost << "options: " << options.ToString();
     ost << "}";
     return ost.str();
   }
 
   AudioOptions options;
 };
 
 struct AudioRecvParameters : RtpParameters<AudioCodec> {
 };
@@ skipping 189 matching lines @@
 };
 
 enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
 
 struct DataSendParameters : RtpSendParameters<DataCodec> {
   std::string ToString() const {
     std::ostringstream ost;
     // Options and extensions aren't used.
     ost << "{";
     ost << "codecs: " << VectorToString(codecs) << ", ";
-    ost << "max_bandwidth_bps: " << max_bandwidth_bps;
+    ost << "max_bitrate_bps: " << max_bitrate_bps;
     ost << "}";
     return ost.str();
   }
 };
 
 struct DataRecvParameters : RtpParameters<DataCodec> {
 };
 
 class DataMediaChannel : public MediaChannel {
  public:
@@ skipping 32 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_