Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
index 5f2eb759c96937cccb48d44f421e4f8a22a4c534..186f4e980016c18399dd216ce93518c605604c8c 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
@@ -327,7 +327,8 @@ 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 g
danilchap
2017/03/16 13:58:02
imagine 30 participant meeting with server-side au
|
return -1; |
csrc_cnames_[SSRC] = c_name; |
@@ -463,8 +464,8 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES( |
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); |
} |