rename timestamps to show epoch
I missed one timestamp in https://webrtc-review.googlesource.com/c/src/+/363946, meaning that the config flag that was added do not yet work for all timestamps in RTCStats objects. The RTCRemoteOutboundRtpStreamStats still has UTC timestamps even if the config flag is set.
I will solve this by saving both an UTC (existing) and env (to be added) timestamp, and then let rtc_stats_collector choose timestamp based on the value of the config flag (just like RTCRemoteInboundRtpStreamStats is done in the 363946 commit).
Before adding the new env_ timestamp I want to make this change. I rename the existing timestamp to show what epoch it uses (NTP or UTC). This will later make it clear which timestamp is which.
So this CL will make no logical change, just renaming members.
I only need to rename the last_sender_report_timestamp_ms, but opted to rename the remote timestamp as well, to be consistent with the naming convention I add in this CL.
Bug: chromium:369369568
Change-Id: Icfe7cf274995b39799e1478a1bb8cdf5134f0b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364782
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43194}
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 269c851..948f198 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -332,10 +332,10 @@
stats.decoding_plc_cng = ds.decoded_plc_cng;
stats.decoding_muted_output = ds.decoded_muted_output;
- stats.last_sender_report_timestamp_ms =
- call_stats.last_sender_report_timestamp_ms;
- stats.last_sender_report_remote_timestamp_ms =
- call_stats.last_sender_report_remote_timestamp_ms;
+ stats.last_sender_report_utc_timestamp_ms =
+ call_stats.last_sender_report_utc_timestamp_ms;
+ stats.last_sender_report_remote_utc_timestamp_ms =
+ call_stats.last_sender_report_remote_utc_timestamp_ms;
stats.sender_reports_packets_sent = call_stats.sender_reports_packets_sent;
stats.sender_reports_bytes_sent = call_stats.sender_reports_bytes_sent;
stats.sender_reports_reports_count = call_stats.sender_reports_reports_count;
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index ca5f916..da58924 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -777,7 +777,7 @@
{
MutexLock lock(&ts_stats_lock_);
- ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+ ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_ntp_timestamp,
last_sr->last_remote_rtp_timestamp);
std::optional<int64_t> remote_to_local_clock_offset =
ntp_estimator_.EstimateRemoteToLocalClockOffset();
@@ -872,11 +872,12 @@
std::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_rtcp_->GetSenderReportStats();
if (rtcp_sr_stats.has_value()) {
- stats.last_sender_report_timestamp_ms =
- rtcp_sr_stats->last_arrival_timestamp.ToMs() -
+ stats.last_sender_report_utc_timestamp_ms =
+ rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
- stats.last_sender_report_remote_timestamp_ms =
- rtcp_sr_stats->last_remote_timestamp.ToMs() - rtc::kNtpJan1970Millisecs;
+ stats.last_sender_report_remote_utc_timestamp_ms =
+ rtcp_sr_stats->last_remote_ntp_timestamp.ToMs() -
+ rtc::kNtpJan1970Millisecs;
stats.sender_reports_packets_sent = rtcp_sr_stats->packets_sent;
stats.sender_reports_bytes_sent = rtcp_sr_stats->bytes_sent;
stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
@@ -1127,8 +1128,8 @@
if (!last_sr.has_value()) {
return std::nullopt;
}
- info.capture_time_ntp_secs = last_sr->last_remote_timestamp.seconds();
- info.capture_time_ntp_frac = last_sr->last_remote_timestamp.fractions();
+ info.capture_time_ntp_secs = last_sr->last_remote_ntp_timestamp.seconds();
+ info.capture_time_ntp_frac = last_sr->last_remote_ntp_timestamp.fractions();
info.capture_time_source_clock = last_sr->last_remote_rtp_timestamp;
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 4e2048d..98da20c 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -67,8 +67,8 @@
// Note that the timestamps below correspond to the time elapsed since the
// Unix epoch.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- std::optional<int64_t> last_sender_report_timestamp_ms;
- std::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_utc_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 62dd4cb..1c64b63 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -237,7 +237,7 @@
{
MutexLock lock(&lock_);
- ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+ ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_ntp_timestamp,
last_sr->last_remote_rtp_timestamp);
}
}
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index d91e68c..655f028 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -100,8 +100,8 @@
std::optional<int64_t> estimated_playout_ntp_timestamp_ms;
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- std::optional<int64_t> last_sender_report_timestamp_ms;
- std::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_utc_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 08ac664..2e97554 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -175,8 +175,8 @@
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- std::optional<int64_t> last_sender_report_timestamp_ms;
- std::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_utc_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
uint32_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 73d2b25..f8c1fed 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -482,8 +482,8 @@
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- std::optional<int64_t> last_sender_report_timestamp_ms;
- std::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_utc_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index a094443..12ee581 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3844,9 +3844,10 @@
}
// remote-outbound-rtp stats.
- info.last_sender_report_timestamp_ms = stats.last_sender_report_timestamp_ms;
- info.last_sender_report_remote_timestamp_ms =
- stats.last_sender_report_remote_timestamp_ms;
+ info.last_sender_report_utc_timestamp_ms =
+ stats.last_sender_report_utc_timestamp_ms;
+ info.last_sender_report_remote_utc_timestamp_ms =
+ stats.last_sender_report_remote_utc_timestamp_ms;
info.sender_reports_packets_sent = stats.sender_reports_packets_sent;
info.sender_reports_bytes_sent = stats.sender_reports_bytes_sent;
info.sender_reports_reports_count = stats.sender_reports_reports_count;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 69f2a3c..f78d1eb 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2697,10 +2697,10 @@
stats.relative_packet_arrival_delay_seconds;
rinfo.interruption_count = stats.interruption_count;
rinfo.total_interruption_duration_ms = stats.total_interruption_duration_ms;
- rinfo.last_sender_report_timestamp_ms =
- stats.last_sender_report_timestamp_ms;
- rinfo.last_sender_report_remote_timestamp_ms =
- stats.last_sender_report_remote_timestamp_ms;
+ rinfo.last_sender_report_utc_timestamp_ms =
+ stats.last_sender_report_utc_timestamp_ms;
+ rinfo.last_sender_report_remote_utc_timestamp_ms =
+ stats.last_sender_report_remote_utc_timestamp_ms;
rinfo.sender_reports_packets_sent = stats.sender_reports_packets_sent;
rinfo.sender_reports_bytes_sent = stats.sender_reports_bytes_sent;
rinfo.sender_reports_reports_count = stats.sender_reports_reports_count;
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index a4ea1b0..4014990 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -253,7 +253,7 @@
void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
MutexLock lock(&rtcp_receiver_lock_);
// New SSRC reset old reports.
- remote_sender_.last_arrival_timestamp.Reset();
+ remote_sender_.last_arrival_ntp_timestamp.Reset();
remote_ssrc_ = ssrc;
}
@@ -356,7 +356,7 @@
std::optional<RtpRtcpInterface::SenderReportStats>
RTCPReceiver::GetSenderReportStats() const {
MutexLock lock(&rtcp_receiver_lock_);
- if (!remote_sender_.last_arrival_timestamp.Valid()) {
+ if (!remote_sender_.last_arrival_ntp_timestamp.Valid()) {
return std::nullopt;
}
@@ -549,9 +549,9 @@
// Only signal that we have received a SR when we accept one.
packet_information->packet_type_flags |= kRtcpSr;
- remote_sender_.last_remote_timestamp = sender_report.ntp();
+ remote_sender_.last_remote_ntp_timestamp = sender_report.ntp();
remote_sender_.last_remote_rtp_timestamp = sender_report.rtp_timestamp();
- remote_sender_.last_arrival_timestamp = env_.clock().CurrentNtpTime();
+ remote_sender_.last_arrival_ntp_timestamp = env_.clock().CurrentNtpTime();
remote_sender_.packets_sent = sender_report.sender_packet_count();
remote_sender_.bytes_sent = sender_report.sender_octet_count();
remote_sender_.reports_count++;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 568b1e2..70f8944 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -301,8 +301,8 @@
if (std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtcp_receiver_.GetSenderReportStats();
last_sr.has_value()) {
- state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
- state.last_rr = last_sr->last_arrival_timestamp;
+ state.remote_sr = CompactNtp(last_sr->last_remote_ntp_timestamp);
+ state.last_rr = last_sr->last_arrival_ntp_timestamp;
}
state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index fe21008..d994075 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -272,8 +272,8 @@
if (std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtcp_receiver_.GetSenderReportStats();
last_sr.has_value()) {
- state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
- state.last_rr = last_sr->last_arrival_timestamp;
+ state.remote_sr = CompactNtp(last_sr->last_remote_ntp_timestamp);
+ state.last_rr = last_sr->last_arrival_ntp_timestamp;
}
state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
index 63224e7..4e1862b 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -811,11 +811,11 @@
auto raw_packet = sr.Build();
receiver_.impl_->IncomingRtcpPacket(raw_packet);
- EXPECT_THAT(
- receiver_.impl_->GetSenderReportStats(),
- Optional(AllOf(Field(&SenderReportStats::last_remote_timestamp, Eq(ntp)),
- Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
- Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+ Optional(AllOf(
+ Field(&SenderReportStats::last_remote_ntp_timestamp, Eq(ntp)),
+ Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
+ Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
}
// Checks that the sender report stats count equals the number of sent RTCP SRs.
@@ -845,7 +845,7 @@
AdvanceTime(kOneWayNetworkDelay);
auto stats = receiver_.impl_->GetSenderReportStats();
ASSERT_THAT(stats, Not(Eq(std::nullopt)));
- EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
+ EXPECT_TRUE(stats->last_arrival_ntp_timestamp.Valid());
}
// Checks that the packet and byte counters from an RTCP SR are not zero once
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index dd69312..a21f9ce 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -660,11 +660,11 @@
sr.SetOctetCount(kOctetCount);
receiver_.impl_->IncomingRtcpPacket(sr.Build());
- EXPECT_THAT(
- receiver_.impl_->GetSenderReportStats(),
- Optional(AllOf(Field(&SenderReportStats::last_remote_timestamp, Eq(ntp)),
- Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
- Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+ Optional(AllOf(
+ Field(&SenderReportStats::last_remote_ntp_timestamp, Eq(ntp)),
+ Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
+ Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
}
// Checks that the remote sender stats count equals the number of sent RTCP SRs.
@@ -691,7 +691,7 @@
ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
auto stats = receiver_.impl_->GetSenderReportStats();
ASSERT_THAT(stats, Not(Eq(std::nullopt)));
- EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
+ EXPECT_TRUE(stats->last_arrival_ntp_timestamp.Valid());
}
// Checks that the packet and byte counters from an RTCP SR are not zero once
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index dd34cf7e..a6bf2df 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -147,9 +147,9 @@
// Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1.
struct SenderReportStats {
// Arrival NTP timestamp for the last received RTCP SR.
- NtpTime last_arrival_timestamp;
+ NtpTime last_arrival_ntp_timestamp;
// Received (a.k.a., remote) NTP timestamp for the last received RTCP SR.
- NtpTime last_remote_timestamp;
+ NtpTime last_remote_ntp_timestamp;
// Received (a.k.a., remote) RTP timestamp from the last received RTCP SR.
uint32_t last_remote_rtp_timestamp = 0;
// Total number of RTP data packets transmitted by the sender since starting
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 29598e0..08f3961 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -522,7 +522,7 @@
cricket::MediaType media_type,
const RTCInboundRtpStreamStats& inbound_audio_stats,
const std::string& transport_id) {
- if (!media_receiver_info.last_sender_report_timestamp_ms.has_value()) {
+ if (!media_receiver_info.last_sender_report_utc_timestamp_ms.has_value()) {
// Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival
// timestamp is not available - i.e., until the first sender report is
// received.
@@ -534,7 +534,8 @@
auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>(
/*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
media_type, media_receiver_info.ssrc()),
- Timestamp::Millis(*media_receiver_info.last_sender_report_timestamp_ms));
+ Timestamp::Millis(
+ *media_receiver_info.last_sender_report_utc_timestamp_ms));
// Populate.
// - RTCRtpStreamStats.
@@ -549,12 +550,12 @@
stats->bytes_sent = media_receiver_info.sender_reports_bytes_sent;
// - RTCRemoteOutboundRtpStreamStats.
stats->local_id = inbound_audio_stats.id();
- // last_sender_report_remote_timestamp_ms is set together with
- // last_sender_report_timestamp_ms.
- RTC_DCHECK(
- media_receiver_info.last_sender_report_remote_timestamp_ms.has_value());
+ // last_sender_report_remote_utc_timestamp_ms is set together with
+ // last_sender_report_utc_timestamp_ms.
+ RTC_DCHECK(media_receiver_info.last_sender_report_remote_utc_timestamp_ms
+ .has_value());
stats->remote_timestamp = static_cast<double>(
- *media_receiver_info.last_sender_report_remote_timestamp_ms);
+ *media_receiver_info.last_sender_report_remote_utc_timestamp_ms);
stats->reports_sent = media_receiver_info.sender_reports_reports_count;
if (media_receiver_info.round_trip_time.has_value()) {
stats->round_trip_time =
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index cdbddf2..fe999ef 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -844,9 +844,9 @@
// remote-outbound-rtp
if (add_remote_outbound_stats) {
graph.remote_outbound_rtp_id = "ROA4";
- media_info.receivers[0].last_sender_report_timestamp_ms =
+ media_info.receivers[0].last_sender_report_utc_timestamp_ms =
kRemoteOutboundStatsTimestampMs;
- media_info.receivers[0].last_sender_report_remote_timestamp_ms =
+ media_info.receivers[0].last_sender_report_remote_utc_timestamp_ms =
kRemoteOutboundStatsRemoteTimestampMs;
media_info.receivers[0].sender_reports_packets_sent =
kRemoteOutboundStatsPacketsSent;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 4604180..ef1dd75 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -382,8 +382,8 @@
if (!last_sr.has_value()) {
return std::nullopt;
}
- info.capture_time_ntp_secs = last_sr->last_remote_timestamp.seconds();
- info.capture_time_ntp_frac = last_sr->last_remote_timestamp.fractions();
+ info.capture_time_ntp_secs = last_sr->last_remote_ntp_timestamp.seconds();
+ info.capture_time_ntp_frac = last_sr->last_remote_ntp_timestamp.fractions();
info.capture_time_source_clock = last_sr->last_remote_rtp_timestamp;
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_) {
@@ -1210,10 +1210,10 @@
return true;
}
int64_t time_since_received = env_.clock().CurrentNtpInMilliseconds() -
- last_sr->last_arrival_timestamp.ToMs();
+ last_sr->last_arrival_ntp_timestamp.ToMs();
// Don't use old SRs to estimate time.
if (time_since_received <= 1) {
- ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+ ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_ntp_timestamp,
last_sr->last_remote_rtp_timestamp);
std::optional<int64_t> remote_to_local_clock_offset =
ntp_estimator_.EstimateRemoteToLocalClockOffset();
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 1dedc3e..eda4f91 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -572,11 +572,12 @@
std::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_video_stream_receiver_.GetSenderReportStats();
if (rtcp_sr_stats) {
- stats.last_sender_report_timestamp_ms =
- rtcp_sr_stats->last_arrival_timestamp.ToMs() -
+ stats.last_sender_report_utc_timestamp_ms =
+ rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
- stats.last_sender_report_remote_timestamp_ms =
- rtcp_sr_stats->last_remote_timestamp.ToMs() - rtc::kNtpJan1970Millisecs;
+ stats.last_sender_report_remote_utc_timestamp_ms =
+ rtcp_sr_stats->last_remote_ntp_timestamp.ToMs() -
+ rtc::kNtpJan1970Millisecs;
stats.sender_reports_packets_sent = rtcp_sr_stats->packets_sent;
stats.sender_reports_bytes_sent = rtcp_sr_stats->bytes_sent;
stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
