Cast: Increase UDP socket send buffer size

media/cast/net/cast_transport_config.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/net/cast_transport_config.h"
 
+#include "net/base/net_errors.h"
+
 namespace media {
 namespace cast {
 
 CastTransportRtpConfig::CastTransportRtpConfig()
     : ssrc(0), feedback_ssrc(0), rtp_payload_type(0) {}
 
 CastTransportRtpConfig::~CastTransportRtpConfig() {}
 
 EncodedFrame::EncodedFrame()
     : dependency(UNKNOWN_DEPENDENCY),
       frame_id(0),
       referenced_frame_id(0),
       rtp_timestamp(0),
       new_playout_delay_ms(0) {}
 
 EncodedFrame::~EncodedFrame() {}
 
 void EncodedFrame::CopyMetadataTo(EncodedFrame* dest) const {
   DCHECK(dest);
   dest->dependency = this->dependency;
   dest->frame_id = this->frame_id;
   dest->referenced_frame_id = this->referenced_frame_id;
   dest->rtp_timestamp = this->rtp_timestamp;
   dest->reference_time = this->reference_time;
 }
 
+int PacketSender::SetSendBufferSize(int32 size) {
+  return net::OK;
+}
+
 RtcpSenderInfo::RtcpSenderInfo()
     : ntp_seconds(0),
       ntp_fraction(0),
       rtp_timestamp(0),
       send_packet_count(0),
       send_octet_count(0) {}
 RtcpSenderInfo::~RtcpSenderInfo() {}
 
 RtcpReportBlock::RtcpReportBlock()
     : remote_ssrc(0),
       media_ssrc(0),
       fraction_lost(0),
       cumulative_lost(0),
       extended_high_sequence_number(0),
       jitter(0),
       last_sr(0),
       delay_since_last_sr(0) {}
 RtcpReportBlock::~RtcpReportBlock() {}
 
 RtcpDlrrReportBlock::RtcpDlrrReportBlock()
     : last_rr(0), delay_since_last_rr(0) {}
 RtcpDlrrReportBlock::~RtcpDlrrReportBlock() {}
 
 }  // namespace cast
 }  // namespace media