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

webrtc/api/webrtcsdp.cc

 /*
  *  Copyright 2011 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 1279 matching lines @@
       transport_info->description.identity_fingerprint.get() : NULL;
 
   // Add the m and c lines.
   InitLine(kLineTypeMedia, type, &os);
   os << " " << port << " " << media_desc->protocol() << fmt;
   std::string mline = os.str();
   UpdateMediaDefaultDestination(candidates, mline, message);
 
   // RFC 4566
   // b=AS:<bandwidth>
-  if (media_desc->bandwidth() >= 1000) {
+  if (media_desc->bandwidth() && (*media_desc->bandwidth() >= 1000)) {
     InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
-    os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
+    os << kSdpDelimiterColon << (*media_desc->bandwidth() / 1000);
     AddLine(os.str(), message);
   }
 
   // Add the a=rtcp line.
   if (IsRtp(media_desc->protocol())) {
     std::string rtcp_line = GetRtcpLine(candidates);
     if (!rtcp_line.empty()) {
       AddLine(rtcp_line, message);
     }
   }
@@ skipping 1238 matching lines @@
     }
 
     // RFC 4566
     // b=* (zero or more bandwidth information lines)
     if (IsLineType(line, kLineTypeSessionBandwidth)) {
       std::string bandwidth;
       if (HasAttribute(line, kApplicationSpecificMaximum)) {
         if (!GetValue(line, kApplicationSpecificMaximum, &bandwidth, error)) {
           return false;
         } else {
-          int b = 0;
-          if (!GetValueFromString(line, bandwidth, &b, error)) {
+          int bandwidth_value = 0;
+          if (!GetValueFromString(line, bandwidth, &bandwidth_value, error)) {
             return false;
           }
+          if (bandwidth_value < 0) {
+            std::ostringstream description;
+            description << "The bandwidth value must not be negative";
+            return ParseFailed(line, description.str(), error);
+          }
+
           // We should never use more than the default bandwidth for RTP-based
           // data channels. Don't allow SDP to set the bandwidth, because
           // that would give JS the opportunity to "break the Internet".
           // See: https://code.google.com/p/chromium/issues/detail?id=280726
           if (media_type == cricket::MEDIA_TYPE_DATA && IsRtp(protocol) &&
-              b > cricket::kDataMaxBandwidth / 1000) {
+              bandwidth_value > cricket::kDataMaxBandwidth / 1000) {
             std::ostringstream description;
             description << "RTP-based data channels may not send more than "
                         << cricket::kDataMaxBandwidth / 1000 << "kbps.";
             return ParseFailed(line, description.str(), error);
           }
-          media_desc->set_bandwidth(b * 1000);
+          media_desc->set_bandwidth(rtc::Optional<int>(1000 * bandwidth_value));
         }
       }
       continue;
     }
 
     if (!IsLineType(line, kLineTypeAttributes)) {
       // TODO: Handle other lines if needed.
       LOG(LS_INFO) << "Ignored line: " << line;
       continue;
     }
@@ skipping 569 matching lines @@
     UpdateCodec<AudioContentDescription, cricket::AudioCodec>(
         media_desc, payload_type, feedback_param);
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     UpdateCodec<VideoContentDescription, cricket::VideoCodec>(
         media_desc, payload_type, feedback_param);
   }
   return true;
 }
 
 }  // namespace webrtc