Drop the RTT as input to IsRetransmitOfOldPacket.
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}diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 49d29f0..bb2833f 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -301,7 +301,7 @@
}
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 @@
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;
}