Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index e9deb3d..b476f2c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -1135,13 +1135,10 @@
id_,
"SendNACK(size:%u)", size);
- // Use RTT from RtcpRttStats class if provided.
- uint16_t rtt = rtt_ms();
- if (rtt == 0) {
- rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
- }
+ uint16_t avg_rtt = 0;
+ rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &avg_rtt, NULL, NULL);
- int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5.
+ int64_t wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5.
if (wait_time == 5) {
wait_time = 100; // During startup we don't have an RTT.
}
@@ -1625,12 +1622,9 @@
nack_sequence_numbers.size() == 0) {
return;
}
- // Use RTT from RtcpRttStats class if provided.
- uint16_t rtt = rtt_ms();
- if (rtt == 0) {
- rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
- }
- rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt);
+ uint16_t avg_rtt = 0;
+ rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &avg_rtt, NULL, NULL);
+ rtp_sender_.OnReceivedNACK(nack_sequence_numbers, avg_rtt);
}
int32_t ModuleRtpRtcpImpl::LastReceivedNTP(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 075770d..054e2f3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -410,7 +410,6 @@
Clock* clock_;
private:
- FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
int64_t RtcpReportInterval();
void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 50f7f2e..6248f49 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -122,13 +122,6 @@
// No RTT from other ssrc.
EXPECT_EQ(-1,
rtp_rtcp_impl_->RTT(kSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
-
- // Verify RTT from rtt_stats config.
- EXPECT_EQ(0U, rtt_stats_.LastProcessedRtt());
- EXPECT_EQ(0U, rtp_rtcp_impl_->rtt_ms());
- rtp_rtcp_impl_->Process();
- EXPECT_EQ(100U, rtt_stats_.LastProcessedRtt());
- EXPECT_EQ(100U, rtp_rtcp_impl_->rtt_ms());
}
TEST_F(RtpRtcpImplTest, SetRtcpXrRrtrStatus) {
@@ -154,6 +147,7 @@
// Verify RTT.
EXPECT_EQ(0U, rtt_stats_.LastProcessedRtt());
EXPECT_EQ(0U, rtp_rtcp_impl_->rtt_ms());
+
rtp_rtcp_impl_->Process();
EXPECT_EQ(100U, rtt_stats_.LastProcessedRtt());
EXPECT_EQ(100U, rtp_rtcp_impl_->rtt_ms());