Fix perf issue when timinig out receiver infos in RTCP.
BUG=b/33270241
Review-Url: https://codereview.webrtc.org/2664163002
Cr-Commit-Position: refs/heads/master@{#16414}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 51c4aa8..59049be 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -145,11 +145,7 @@
int64_t RTCPReceiver::LastReceivedReceiverReport() const {
rtc::CritScope lock(&rtcp_receiver_lock_);
- int64_t last_received_rr = -1;
- for (const auto& kv : received_infos_)
- if (kv.second.last_time_received_ms > last_received_rr)
- last_received_rr = kv.second.last_time_received_ms;
- return last_received_rr;
+ return last_received_rr_ms_;
}
void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
@@ -445,6 +441,8 @@
return;
}
+ last_received_rr_ms_ = clock_->TimeInMilliseconds();
+
const uint32_t remote_ssrc = receiver_report.sender_ssrc();
packet_information->remote_ssrc = remote_ssrc;
@@ -478,8 +476,6 @@
ReportBlockWithRtt* report_block_info =
&received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
-
- last_received_rr_ms_ = clock_->TimeInMilliseconds();
report_block_info->report_block.remoteSSRC = remote_ssrc;
report_block_info->report_block.sourceSSRC = report_block.source_ssrc();
report_block_info->report_block.fractionLost = report_block.fraction_lost();
@@ -489,7 +485,7 @@
report_block_info->report_block.extendedHighSeqNum) {
// We have successfully delivered new RTP packets to the remote side after
// the last RR was sent from the remote side.
- last_increased_sequence_number_ms_ = last_received_rr_ms_;
+ last_increased_sequence_number_ms_ = clock_->TimeInMilliseconds();
}
report_block_info->report_block.extendedHighSeqNum =
report_block.extended_high_seq_num();
@@ -579,22 +575,30 @@
bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
rtc::CritScope lock(&rtcp_receiver_lock_);
- bool update_bounding_set = false;
int64_t now_ms = clock_->TimeInMilliseconds();
// Use audio define since we don't know what interval the remote peer use.
- int64_t timeouted_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
+ int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
+ if (oldest_received_info_ms_ >= timeout_ms)
+ return false;
+
+ bool update_bounding_set = false;
+ oldest_received_info_ms_ = -1;
for (auto receive_info_it = received_infos_.begin();
receive_info_it != received_infos_.end();) {
ReceiveInformation* receive_info = &receive_info_it->second;
if (receive_info->last_time_received_ms > 0) {
- if (receive_info->last_time_received_ms < timeouted_ms) {
+ if (receive_info->last_time_received_ms < timeout_ms) {
// No rtcp packet for the last 5 regular intervals, reset limitations.
receive_info->tmmbr.clear();
// Prevent that we call this over and over again.
receive_info->last_time_received_ms = 0;
// Send new TMMBN to all channels using the default codec.
update_bounding_set = true;
+ } else if (oldest_received_info_ms_ == -1 ||
+ receive_info->last_time_received_ms <
+ oldest_received_info_ms_) {
+ oldest_received_info_ms_ = receive_info->last_time_received_ms;
}
++receive_info_it;
} else if (receive_info->ready_for_delete) {
@@ -1070,11 +1074,11 @@
int64_t now_ms = clock_->TimeInMilliseconds();
// Use audio define since we don't know what interval the remote peer use.
- int64_t timeouted_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
+ int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
for (auto& kv : received_infos_) {
for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
- if (it->second.last_updated_ms < timeouted_ms) {
+ if (it->second.last_updated_ms < timeout_ms) {
// Erase timeout entries.
it = kv.second.tmmbr.erase(it);
} else {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index b67d07f..1e4d88b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -244,11 +244,12 @@
// Received report blocks.
ReportBlockMap received_report_blocks_ GUARDED_BY(rtcp_receiver_lock_);
ReceivedInfoMap received_infos_ GUARDED_BY(rtcp_receiver_lock_);
+ int64_t oldest_received_info_ms_ GUARDED_BY(rtcp_receiver_lock_);
std::map<uint32_t, std::string> received_cnames_
GUARDED_BY(rtcp_receiver_lock_);
// The last time we received an RTCP RR.
- int64_t last_received_rr_ms_;
+ int64_t last_received_rr_ms_ GUARDED_BY(rtcp_receiver_lock_);
// The time we last received an RTCP RR telling we have successfully
// delivered RTP packet to the remote side.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3c986e9..e0bcebd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -211,7 +211,7 @@
if (rtcp_sender_.TimeToSendRTCPReport())
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
- if (UpdateRTCPReceiveInformationTimers()) {
+ if (TMMBR() && rtcp_receiver_.UpdateRTCPReceiveInformationTimers()) {
// A receiver has timed out.
rtcp_receiver_.UpdateTmmbr();
}
@@ -867,12 +867,6 @@
return true;
}
-bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
- // If this returns true this channel has timed out.
- // Periodically check if this is true and if so call UpdateTMMBR.
- return rtcp_receiver_.UpdateRTCPReceiveInformationTimers();
-}
-
// Called from RTCPsender.
std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
return rtcp_receiver_.BoundingSet(tmmbr_owner);
