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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

Issue 1491843004: [rtp_rtcp] RTT intermidiate calculation use ntp time instead of milliseconds. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 10 months ago
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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 8ef4f619ff03959d21c81ce58967e474fc022e8c..a3de360cf6a4a399eb82bbfa5353bf2e83061d2a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -195,8 +195,6 @@ RTCPSender::RTCPSender(
xr_send_receiver_reference_time_enabled_(false),
packet_type_counter_observer_(packet_type_counter_observer) {
- memset(last_send_report_, 0, sizeof(last_send_report_));
- memset(last_rtcp_time_, 0, sizeof(last_rtcp_time_));
RTC_DCHECK(transport_ != nullptr);
builders_[kRtcpSr] = &RTCPSender::BuildSR;
@@ -445,46 +443,7 @@ bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
return false;
}
-int64_t RTCPSender::SendTimeOfSendReport(uint32_t sendReport) {
- CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
-
- // This is only saved when we are the sender
- if ((last_send_report_[0] == 0) || (sendReport == 0)) {
- return 0; // will be ignored
- } else {
- for (int i = 0; i < RTCP_NUMBER_OF_SR; ++i) {
- if (last_send_report_[i] == sendReport)
- return last_rtcp_time_[i];
- }
- }
- return 0;
-}
-
-bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp,
- int64_t* time_ms) const {
- CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
-
- if (last_xr_rr_.empty()) {
- return false;
- }
- std::map<uint32_t, int64_t>::const_iterator it = last_xr_rr_.find(mid_ntp);
- if (it == last_xr_rr_.end()) {
- return false;
- }
- *time_ms = it->second;
- return true;
-}
-
rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
- for (int i = (RTCP_NUMBER_OF_SR - 2); i >= 0; i--) {
- // shift old
- last_send_report_[i + 1] = last_send_report_[i];
- last_rtcp_time_[i + 1] = last_rtcp_time_[i];
- }
-
- last_rtcp_time_[0] = Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_);
- last_send_report_[0] = (ctx.ntp_sec_ << 16) + (ctx.ntp_frac_ >> 16);
-
// The timestamp of this RTCP packet should be estimated as the timestamp of
// the frame being captured at this moment. We are calculating that
// timestamp as the last frame's timestamp + the time since the last frame
@@ -752,11 +711,6 @@ rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildBYE(const RtcpContext& ctx) {
rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildReceiverReferenceTime(
const RtcpContext& ctx) {
- if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR)
- last_xr_rr_.erase(last_xr_rr_.begin());
- last_xr_rr_.insert(std::pair<uint32_t, int64_t>(
- RTCPUtility::MidNtp(ctx.ntp_sec_, ctx.ntp_frac_),
- Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_)));
rtcp::ExtendedReports* xr = new rtcp::ExtendedReports();
xr->From(ssrc_);
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