Cast: Don't cancel re-transmission for some cases

media/cast/transport/rtp_sender/rtp_sender.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 // This file contains the interface to the cast RTP sender.
 
 #ifndef MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
 #define MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
 
 #include <map>
@@ skipping 32 matching lines @@
   // Initialize audio stack. Audio must be initialized prior to sending encoded
   // audio frames. Returns false if configuration is invalid.
   bool InitializeAudio(const CastTransportAudioConfig& config);
 
   // Initialize video stack. Video must be initialized prior to sending encoded
   // video frames. Returns false if configuration is invalid.
   bool InitializeVideo(const CastTransportVideoConfig& config);
 
   void SendFrame(const EncodedFrame& frame);
 
-  void ResendPackets(const MissingFramesAndPacketsMap& missing_packets);
+  void ResendPackets(const MissingFramesAndPacketsMap& missing_packets,
+                     bool cancel_rtx_if_not_in_list);
 
   size_t send_packet_count() const {
     return packetizer_ ? packetizer_->send_packet_count() : 0;
   }
   size_t send_octet_count() const {
     return packetizer_ ? packetizer_->send_octet_count() : 0;
   }
   uint32 ssrc() const { return config_.ssrc; }
 
  private:
@@ skipping 10 matching lines @@
   base::WeakPtrFactory<RtpSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpSender);
 };
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_