Fix rtcp_sender to support sdes with 31 chunk

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.
  */
 
@@ skipping 309 matching lines @@
   RTC_DCHECK_LT(strlen(c_name), RTCP_CNAME_SIZE);
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   cname_ = c_name;
   return 0;
 }
 
 int32_t RTCPSender::AddMixedCNAME(uint32_t SSRC, const char* c_name) {
   RTC_DCHECK(c_name);
   RTC_DCHECK_LT(strlen(c_name), RTCP_CNAME_SIZE);
   rtc::CritScope lock(&critical_section_rtcp_sender_);
-  if (csrc_cnames_.size() >= kRtpCsrcSize)
+  // One spot is reserved for ssrc_/cname_.
+  if (csrc_cnames_.size() >= rtcp::Sdes::kMaxNumberOfChunks - 1)

[nisse-webrtc, 2017/03/16 13:42:44]

I'd need a bit more context to understand what's going on. 

I've had a quick look at RFC 3550. So kRtpCsrsSize == 15 is the maximum number
of csrs in an RTP header, i.e., a limit of the number of streams that can be
mixed together in a single RTP packet.

SDES is the source description RTCP message (RFC3550), and there can be at most
31 chunks, of which 30 can be CSRSs (and you can have even more, put into
additional SDES packets). Why would one have a larger number here? We have 50
sources to describe, even though at any point in time the mixer includes only a
few of them?

And what are we using the cname attribute for?

[danilchap, 2017/03/16 13:58:02]

imagine 30 participant meeting with server-side audio mixing.
Sdes list all 30 participiant, and each rtp packet include csrcs of the
currently speaking people (thus it need place for only few csrcs). Using csrcs
client can show who was actually speaking.
cname is a way to identify the participant. same cname might be mapped to
several srcs (e.g. 1 audio, 3 video ssrcs).

     return -1;
 
   csrc_cnames_[SSRC] = c_name;
   return 0;
 }
 
 int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   auto it = csrc_cnames_.find(SSRC);
 
@@ skipping 115 matching lines @@
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES(
     const RtcpContext& ctx) {
   size_t length_cname = cname_.length();
   RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);
 
   rtcp::Sdes* sdes = new rtcp::Sdes();
   sdes->AddCName(ssrc_, cname_);
 
-  for (const auto it : csrc_cnames_)
-    sdes->AddCName(it.first, it.second);
+  for (const auto& it : csrc_cnames_)
+    RTC_CHECK(sdes->AddCName(it.first, it.second));
 
   return std::unique_ptr<rtcp::RtcpPacket>(sdes);
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) {
   rtcp::ReceiverReport* report = new rtcp::ReceiverReport();
   report->SetSenderSsrc(ssrc_);
   for (auto it : report_blocks_)
     report->AddReportBlock(it.second);
 
@@ skipping 522 matching lines @@
     max_packet_size = max_packet_size_;
   }
 
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_packet_size, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc