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Drop the RTT as input to IsRetransmitOfOldPacket.

Browse Source 
Bug: webrtc:7135
Change-Id: I532334934a757ba0ea6a2daf97b0f1cfd04246e6
Reviewed-on: https://webrtc-review.googlesource.com/12320
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23366}
This commit is contained in:
Niels Möller
2018-05-23 13:54:51 +02:00
committed by Commit Bot
parent 89ee4a6c8c
commit eda0087e57
5 changed files with 14 additions and 24 deletions
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23
modules/rtp_rtcp/source/receive_statistics_impl.cc
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@ -301,7 +301,7 @@ uint32_t StreamStatisticianImpl::BitrateReceived() const {
}
bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
const RTPHeader& header, int64_t min_rtt) const {
const RTPHeader& header) const {
rtc::CritScope cs(&stream_lock_);
if (InOrderPacketInternal(header.sequenceNumber)) {
return false;
@ -317,20 +317,17 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
int64_t max_delay_ms = 0;
if (min_rtt == 0) {
// Jitter standard deviation in samples.
float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
// 2 times the standard deviation => 95% confidence.
// And transform to milliseconds by dividing by the frequency in kHz.
max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
// Jitter standard deviation in samples.
float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
// Min max_delay_ms is 1.
if (max_delay_ms == 0) {
max_delay_ms = 1;
}
} else {
max_delay_ms = (min_rtt / 3) + 1;
// 2 times the standard deviation => 95% confidence.
// And transform to milliseconds by dividing by the frequency in kHz.
max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
// Min max_delay_ms is 1.
if (max_delay_ms == 0) {
max_delay_ms = 1;
}
return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
}