Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 9ceebd99e02257027ebf137c7dfd59b8054e89ad..61229245118b3255365428a6584a91c064d93cfd 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -1042,23 +1042,6 @@ bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet, |
return true; |
} |
-void RTPSender::SetSendingStatus(bool enabled) { |
- if (!enabled) { |
- rtc::CritScope lock(&send_critsect_); |
- if (!ssrc_forced_) { |
- // Generate a new SSRC. |
- ssrc_db_->ReturnSSRC(ssrc_); |
- ssrc_ = ssrc_db_->CreateSSRC(); |
- RTC_DCHECK(ssrc_ != 0); |
- } |
- // Don't initialize seq number if SSRC passed externally. |
- if (!sequence_number_forced_ && !ssrc_forced_) { |
- // Generate a new sequence number. |
- sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
- } |
- } |
-} |
- |
void RTPSender::SetSendingMediaStatus(bool enabled) { |
rtc::CritScope lock(&send_critsect_); |
sending_media_ = enabled; |