zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Delete unused functions in RtpSender, RtcpSender and RtcpReceiver

Browse Source 
These functions are not longer used by the RtpRtcp implementations.

Bug: None
Change-Id: Ibc36433b253b264de4cdcdf380f5ec1df201b17a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207862
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33282}
This commit is contained in:
Danil Chapovalov
2021-02-16 11:59:29 +01:00
committed by Commit Bot
parent f4e3e2b83f
commit 3562318bde
9 changed files with 91 additions and 261 deletions
Download Patch File Download Diff File Expand all files Collapse all files

38
modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -273,15 +273,6 @@ bool RTCPSender::TMMBR() const {
return IsFlagPresent(RTCPPacketType::kRtcpTmmbr);
}
void RTCPSender::SetTMMBRStatus(bool enable) {
MutexLock lock(&mutex_rtcp_sender_);
if (enable) {
SetFlag(RTCPPacketType::kRtcpTmmbr, false);
} else {
ConsumeFlag(RTCPPacketType::kRtcpTmmbr, true);
}
}
void RTCPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
MutexLock lock(&mutex_rtcp_sender_);
max_packet_size_ = max_packet_size;
@ -330,31 +321,6 @@ int32_t RTCPSender::SetCNAME(const char* 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);
MutexLock lock(&mutex_rtcp_sender_);
// One spot is reserved for ssrc_/cname_.
// TODO(danilchap): Add support for more than 30 contributes by sending
// several sdes packets.
if (csrc_cnames_.size() >= rtcp::Sdes::kMaxNumberOfChunks - 1)
return -1;
csrc_cnames_[SSRC] = c_name;
return 0;
}
int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
MutexLock lock(&mutex_rtcp_sender_);
auto it = csrc_cnames_.find(SSRC);
if (it == csrc_cnames_.end())
return -1;
csrc_cnames_.erase(it);
return 0;
}
bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
/*
For audio we use a configurable interval (default: 5 seconds)
@ -465,10 +431,6 @@ void RTCPSender::BuildSDES(const RtcpContext& ctx, PacketSender& sender) {
rtcp::Sdes sdes;
sdes.AddCName(ssrc_, cname_);
for (const auto& it : csrc_cnames_)
RTC_CHECK(sdes.AddCName(it.first, it.second));
sender.AppendPacket(sdes);
}