[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <assert.h>  // assert
 #include <string.h>  // memcpy
 
 #include <algorithm>  // min
 #include <limits>     // max
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 using RTCPUtility::RTCPCnameInformation;
 
 NACKStringBuilder::NACKStringBuilder()
     : stream_(""), count_(0), prevNack_(0), consecutive_(false) {}
@@ skipping 427 matching lines @@
   // the frame being captured at this moment. We are calculating that
   // timestamp as the last frame's timestamp + the time since the last frame
   // was captured.
   uint32_t rtp_timestamp =
       start_timestamp_ + last_rtp_timestamp_ +
       (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
           (ctx.feedback_state_.frequency_hz / 1000);
 
   rtcp::SenderReport* report = new rtcp::SenderReport();
   report->From(ssrc_);
-  report->WithNtpSec(ctx.ntp_sec_);
-  report->WithNtpFrac(ctx.ntp_frac_);
+  report->WithNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
   report->WithRtpTimestamp(rtp_timestamp);
   report->WithPacketCount(ctx.feedback_state_.packets_sent);
   report->WithOctetCount(ctx.feedback_state_.media_bytes_sent);
 
   for (auto it : report_blocks_)
     report->WithReportBlock(it.second);
 
   report_blocks_.clear();
 
   return rtc::scoped_ptr<rtcp::SenderReport>(report);
@@ skipping 562 matching lines @@
     Transport* const transport_;
     bool send_failure_;
   } sender(transport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc