Cast: Remove unused RTCP messages

media/cast/sender/audio_sender.cc

 // Copyright 2014 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.
 
 #include "media/cast/sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/cast_defines.h"
@@ skipping 56 matching lines @@
                                     weak_factory_.GetWeakPtr())));
     cast_initialization_status_ = audio_encoder_->InitializationResult();
   } else {
     NOTREACHED();  // No support for external audio encoding.
     cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
   }
 
   media::cast::CastTransportRtpConfig transport_config;
   transport_config.ssrc = audio_config.ssrc;
   transport_config.feedback_ssrc = audio_config.incoming_feedback_ssrc;
-  transport_config.c_name = audio_config.rtcp_c_name;
   transport_config.rtp_payload_type = audio_config.rtp_payload_type;
   // TODO(miu): AudioSender needs to be like VideoSender in providing an upper
   // limit on the number of in-flight frames.
   transport_config.stored_frames = max_unacked_frames_;
   transport_config.aes_key = audio_config.aes_key;
   transport_config.aes_iv_mask = audio_config.aes_iv_mask;
 
   transport_sender->InitializeAudio(
       transport_config,
       base::Bind(&AudioSender::OnReceivedCastFeedback,
@@ skipping 198 matching lines @@
   last_send_time_ = cast_environment_->Clock()->NowTicks();
 
   // Sending this extra packet is to kick-start the session. There is
   // no need to optimize re-transmission for this case.
   transport_sender_->ResendPackets(
       true, missing_frames_and_packets, false, min_rtt_);
 }
 
 }  // namespace cast
 }  // namespace media