Send Sdes using RtcpPacket

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

 /*
  *  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.
  */
 
@@ skipping 80 matching lines @@
  void SetStartTimestamp(uint32_t start_timestamp);
 
  void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
 
  void SetSSRC(uint32_t ssrc);
 
  void SetRemoteSSRC(uint32_t ssrc);
 
  int32_t SetCNAME(const char* cName);
 
- int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]);
+ int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
 
  int32_t RemoveMixedCNAME(uint32_t SSRC);
 
  int64_t SendTimeOfSendReport(uint32_t sendReport);
 
  bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
 
  bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
 
  int32_t SendRTCP(const FeedbackState& feedback_state,
@@ skipping 81 matching lines @@
                  uint64_t pictureID,
                  uint8_t* rtcp_buffer,
                  int buffer_size);
 
  BuildResult BuildSR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildRR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildExtendedJitterReport(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildSDEC(RtcpContext* context)
+ BuildResult BuildSDES(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildPLI(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildREMB(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildTMMBR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildTMMBN(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildAPP(RtcpContext* context)
@@ skipping 31 matching lines @@
  bool extended_jitter_report_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
 
  int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
 
  uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
  uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
  int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
  uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
  // SSRC that we receive on our RTP channel
  uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
- char cname_[RTCP_CNAME_SIZE] GUARDED_BY(critical_section_rtcp_sender_);
+ std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
 
  ReceiveStatistics* receive_statistics_
      GUARDED_BY(critical_section_rtcp_sender_);
  std::map<uint32_t, rtcp::ReportBlock> report_blocks_
      GUARDED_BY(critical_section_rtcp_sender_);
- // TODO(sprang): Can we avoid pointers here?
- std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> csrc_cnames_
+ std::map<uint32_t, std::string> csrc_cnames_
      GUARDED_BY(critical_section_rtcp_sender_);
 
  // Sent
  uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY(
      critical_section_rtcp_sender_);  // allow packet loss and RTT above 1 sec
  int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY(
      critical_section_rtcp_sender_);
 
  // Sent XR receiver reference time report.
  // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
@@ skipping 55 matching lines @@
  std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
 
  typedef BuildResult (RTCPSender::*Builder)(RtcpContext*);
  std::map<RTCPPacketType, Builder> builders_;
 
  class PacketBuiltCallback;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_