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Add timeout for REMB bandwidth estimates.

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BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/458004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1925 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2012-03-23 01:23:55 +00:00
parent 270bd9aaa0
commit 710401bab2
2 changed files with 27 additions and 11 deletions
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33
src/video_engine/vie_remb.cc
View File

@ -22,6 +22,7 @@
namespace webrtc {
const int kRembSendIntervallMs = 1000;
const int kRembTimeOutThresholdMs = 2000;
// % threshold for if we should send a new REMB asap.
const int kSendThresholdPercent = 97;
@ -68,7 +69,7 @@ void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
break;
}
}
bitrates_.erase(ssrc);
update_time_bitrates_.erase(ssrc);
}
void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
@ -144,17 +145,20 @@ void VieRemb::OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate) {
CriticalSectionScoped cs(list_crit_.get());
// Check if this is a new ssrc and add it to the map if it is.
if (bitrates_.find(ssrc) == bitrates_.end()) {
bitrates_[ssrc] = bitrate;
if (update_time_bitrates_.find(ssrc) == update_time_bitrates_.end()) {
update_time_bitrates_[ssrc] = std::make_pair(
TickTime::MillisecondTimestamp(), bitrate);
}
int new_remb_bitrate = last_send_bitrate_ - bitrates_[ssrc] + bitrate;
int new_remb_bitrate = last_send_bitrate_ -
update_time_bitrates_[ssrc].second + bitrate;
if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
// The new bitrate estimate is less than kSendThresholdPercent % of the last
// report. Send a REMB asap.
last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervallMs;
}
bitrates_[ssrc] = bitrate;
update_time_bitrates_[ssrc] = std::make_pair(
TickTime::MillisecondTimestamp(), bitrate);
}
void VieRemb::OnReceivedRemb(unsigned int bitrate) {
@ -191,7 +195,7 @@ WebRtc_Word32 VieRemb::Process() {
// Calculate total receive bitrate estimate.
list_crit_->Enter();
int total_bitrate = 0;
int num_bitrates = bitrates_.size();
int num_bitrates = update_time_bitrates_.size();
if (num_bitrates == 0) {
list_crit_->Leave();
@ -201,10 +205,21 @@ WebRtc_Word32 VieRemb::Process() {
// TODO(mflodman) Use std::vector and change RTP module API.
unsigned int* ssrcs = new unsigned int[num_bitrates];
// Remove any timed out estimates.
SsrcTimeBitrate::iterator it = update_time_bitrates_.begin();
while (it != update_time_bitrates_.end()) {
if (TickTime::MillisecondTimestamp() - it->second.first >
kRembTimeOutThresholdMs) {
update_time_bitrates_.erase(it++);
} else {
++it;
}
}
int idx = 0;
for (SsrcBitrate::iterator it = bitrates_.begin(); it != bitrates_.end();
++it, ++idx) {
total_bitrate += it->second;
for (it = update_time_bitrates_.begin(); it != update_time_bitrates_.end();
++it, ++idx) {
total_bitrate += it->second.second;
ssrcs[idx] = it->first;
}

5
src/video_engine/vie_remb.h
View File

@ -76,7 +76,8 @@ class VieRemb : public RtpRemoteBitrateObserver, public Module {
private:
typedef std::list<RtpRtcp*> RtpModules;
typedef std::map<unsigned int, unsigned int> SsrcBitrate;
typedef std::map<unsigned int, std::pair<int64_t, unsigned int> >
SsrcTimeBitrate;
ProcessThread* process_thread_;
scoped_ptr<CriticalSectionWrapper> list_crit_;
@ -95,7 +96,7 @@ class VieRemb : public RtpRemoteBitrateObserver, public Module {
RtpModules rtcp_sender_;
// The last bitrate update for each SSRC.
SsrcBitrate bitrates_;
SsrcTimeBitrate update_time_bitrates_;
};
} // namespace webrtc