Use environment monotonic timestamps (i.e. not UTC) in RTCStats.
Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 8e2893a..2578523 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -351,6 +351,13 @@
bool dscp() const { return media_config.enable_dscp; }
void set_dscp(bool enable) { media_config.enable_dscp = enable; }
+ bool stats_timestamp_with_environment_clock() const {
+ return media_config.stats_timestamp_with_environment_clock;
+ }
+ void set_stats_timestamp_with_environment_clock(bool enable) {
+ media_config.stats_timestamp_with_environment_clock = enable;
+ }
+
bool cpu_adaptation() const {
return media_config.video.enable_cpu_adaptation;
}
diff --git a/media/base/media_config.h b/media/base/media_config.h
index 7827705..45fd903 100644
--- a/media/base/media_config.h
+++ b/media/base/media_config.h
@@ -22,6 +22,17 @@
// and delete this flag.
bool enable_dscp = true;
+ // If true, RTCStats timestamps are sourced from the monotonically increasing
+ // environment Clock, where the epoch is unspecified (i.e. up to the Clock
+ // implementation). If false, RTCStats timestamps are either sourced from
+ // system clock via rtc::TimeUTCMicros() which is relative to 1970 but not
+ // necessarily monotonically increasing, or from a monotonic clock that is
+ // set to rtc::TimeUTCMicros() at first call, and then procceeds to increase
+ // monotonically.
+ // TODO(webrtc:370535296): Change default value to true and delete this flag
+ // once downstream projects have migrated.
+ bool stats_timestamp_with_environment_clock = false;
+
// Video-specific config.
struct Video {
// Enable WebRTC CPU Overuse Detection. This flag comes from the
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 50917c1..cc92fcd 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -6035,7 +6035,8 @@
report_block.SetCumulativeLost(17);
report_block.SetFractionLost(18);
webrtc::ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, webrtc::Timestamp::Zero());
+ report_block_data.SetReportBlock(0, report_block, webrtc::Timestamp::Zero(),
+ webrtc::Timestamp::Zero());
report_block_data.AddRoundTripTimeSample(webrtc::TimeDelta::Millis(19));
substream.report_block_data = report_block_data;
substream.encode_frame_rate = 20.0;
@@ -6159,7 +6160,8 @@
report_block.SetCumulativeLost(17);
report_block.SetFractionLost(18);
webrtc::ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, webrtc::Timestamp::Zero());
+ report_block_data.SetReportBlock(0, report_block, webrtc::Timestamp::Zero(),
+ webrtc::Timestamp::Zero());
report_block_data.AddRoundTripTimeSample(webrtc::TimeDelta::Millis(19));
substream.report_block_data = report_block_data;
substream.encode_frame_rate = 20.0;
diff --git a/modules/rtp_rtcp/include/report_block_data.cc b/modules/rtp_rtcp/include/report_block_data.cc
index 0d4fed0..84fbea3 100644
--- a/modules/rtp_rtcp/include/report_block_data.cc
+++ b/modules/rtp_rtcp/include/report_block_data.cc
@@ -21,9 +21,12 @@
return TimeDelta::Seconds(jitter()) / rtp_clock_rate_hz;
}
+// TODO(webrtc:370535296): When (webrtc:370535296) is fixed, we don't need the
+// utc timestamp.
void ReportBlockData::SetReportBlock(uint32_t sender_ssrc,
const rtcp::ReportBlock& report_block,
- Timestamp report_block_timestamp_utc) {
+ Timestamp report_block_timestamp_utc,
+ Timestamp report_block_timestamp) {
sender_ssrc_ = sender_ssrc;
source_ssrc_ = report_block.source_ssrc();
fraction_lost_raw_ = report_block.fraction_lost();
@@ -31,6 +34,7 @@
extended_highest_sequence_number_ = report_block.extended_high_seq_num();
jitter_ = report_block.jitter();
report_block_timestamp_utc_ = report_block_timestamp_utc;
+ report_block_timestamp_ = report_block_timestamp;
}
void ReportBlockData::AddRoundTripTimeSample(TimeDelta rtt) {
diff --git a/modules/rtp_rtcp/include/report_block_data.h b/modules/rtp_rtcp/include/report_block_data.h
index f6a23b1..5861f21 100644
--- a/modules/rtp_rtcp/include/report_block_data.h
+++ b/modules/rtp_rtcp/include/report_block_data.h
@@ -72,10 +72,15 @@
TimeDelta jitter(int rtp_clock_rate_hz) const;
// Time in utc epoch (Jan 1st, 1970) the report block was received.
+ // TODO(webrtc:370535296): When (webrtc:370535296) is fixed, we don't need the
+ // utc timestamp.
Timestamp report_block_timestamp_utc() const {
return report_block_timestamp_utc_;
}
+ // Monotonic time when the report block was received.
+ Timestamp report_block_timestamp() const { return report_block_timestamp_; }
+
// Round Trip Time measurments for given (sender_ssrc, source_ssrc) pair.
// Min, max, sum, number of measurements are since beginning of the call.
TimeDelta last_rtt() const { return last_rtt_; }
@@ -91,13 +96,19 @@
extended_highest_sequence_number_ = sn;
}
void set_jitter(uint32_t jitter) { jitter_ = jitter; }
+ // TODO(webrtc:370535296): When (webrtc:370535296) is fixed, we don't need the
+ // utc timestamp.
void set_report_block_timestamp_utc(Timestamp arrival_time) {
report_block_timestamp_utc_ = arrival_time;
}
+ void set_report_block_timestamp(Timestamp arrival_time) {
+ report_block_timestamp_ = arrival_time;
+ }
void SetReportBlock(uint32_t sender_ssrc,
const rtcp::ReportBlock& report_block,
- Timestamp report_block_timestamp_utc);
+ Timestamp report_block_timestamp_utc,
+ Timestamp report_block_timestamp);
void AddRoundTripTimeSample(TimeDelta rtt);
private:
@@ -107,7 +118,10 @@
int32_t cumulative_lost_ = 0;
uint32_t extended_highest_sequence_number_ = 0;
uint32_t jitter_ = 0;
+ // TODO(webrtc:370535296): When (webrtc:370535296) is fixed, we don't need the
+ // utc timestamp.
Timestamp report_block_timestamp_utc_ = Timestamp::Zero();
+ Timestamp report_block_timestamp_ = Timestamp::Zero();
TimeDelta last_rtt_ = TimeDelta::Zero();
TimeDelta sum_rtt_ = TimeDelta::Zero();
size_t num_rtts_ = 0;
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 6116470..9e5a2e1 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -624,7 +624,7 @@
// https://tools.ietf.org/html/rfc868).
report_block_data->SetReportBlock(
remote_ssrc, report_block,
- Timestamp::Millis(now_ntp.ToMs() - rtc::kNtpJan1970Millisecs));
+ Timestamp::Millis(now_ntp.ToMs() - rtc::kNtpJan1970Millisecs), now);
uint32_t send_time_ntp = report_block.last_sr();
// RFC3550, section 6.4.1, LSR field discription states:
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
index f07d765..858b580 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
@@ -375,7 +375,7 @@
auto sender_it = local_senders_by_ssrc_.find(block.source_ssrc());
if (sender_it != local_senders_by_ssrc_.end()) {
LocalSenderState& state = *sender_it->second;
- state.report_block.SetReportBlock(sender_ssrc, block, now_utc);
+ state.report_block.SetReportBlock(sender_ssrc, block, now_utc, now);
if (rtt.has_value()) {
state.report_block.AddRoundTripTimeSample(*rtt);
}
@@ -385,7 +385,7 @@
// No registered sender for this report block, still report it to the
// network link.
ReportBlockData report_block;
- report_block.SetReportBlock(sender_ssrc, block, now_utc);
+ report_block.SetReportBlock(sender_ssrc, block, now_utc, now);
if (rtt.has_value()) {
report_block.AddRoundTripTimeSample(*rtt);
}
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index c4546a9..1171bc8 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -762,7 +762,7 @@
configuration_ = configuration;
legacy_stats_ = std::make_unique<LegacyStatsCollector>(this);
- stats_collector_ = RTCStatsCollector::Create(this);
+ stats_collector_ = RTCStatsCollector::Create(this, env_);
sdp_handler_ =
SdpOfferAnswerHandler::Create(this, configuration, dependencies,
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 2375dad..29598e0 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -843,14 +843,18 @@
const ReportBlockData& report_block,
cricket::MediaType media_type,
const std::map<std::string, RTCOutboundRtpStreamStats*>& outbound_rtps,
- const RTCStatsReport& report) {
+ const RTCStatsReport& report,
+ const bool stats_timestamp_with_environment_clock) {
// RTCStats' timestamp generally refers to when the metric was sampled, but
// for "remote-[outbound/inbound]-rtp" it refers to the local time when the
// Report Block was received.
+ Timestamp arrival_timestamp = stats_timestamp_with_environment_clock
+ ? report_block.report_block_timestamp()
+ : report_block.report_block_timestamp_utc();
auto remote_inbound = std::make_unique<RTCRemoteInboundRtpStreamStats>(
RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc(
media_type, report_block.source_ssrc()),
- report_block.report_block_timestamp_utc());
+ arrival_timestamp);
remote_inbound->ssrc = report_block.source_ssrc();
remote_inbound->kind =
media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video";
@@ -1114,13 +1118,18 @@
rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
PeerConnectionInternal* pc,
+ const Environment& env,
int64_t cache_lifetime_us) {
- return rtc::make_ref_counted<RTCStatsCollector>(pc, cache_lifetime_us);
+ return rtc::make_ref_counted<RTCStatsCollector>(pc, env, cache_lifetime_us);
}
RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc,
+ const Environment& env,
int64_t cache_lifetime_us)
: pc_(pc),
+ env_(env),
+ stats_timestamp_with_environment_clock_(
+ pc->GetConfiguration().stats_timestamp_with_environment_clock()),
signaling_thread_(pc->signaling_thread()),
worker_thread_(pc->worker_thread()),
network_thread_(pc->network_thread()),
@@ -1179,10 +1188,16 @@
// case of already gathering stats, `callback_` will be invoked when there
// are no more pending partial reports.
- // "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970,
- // UTC), in microseconds. The system clock could be modified and is not
- // necessarily monotonically increasing.
- Timestamp timestamp = Timestamp::Micros(rtc::TimeUTCMicros());
+ Timestamp timestamp =
+ stats_timestamp_with_environment_clock_
+ ?
+ // "Now" using a monotonically increasing timer.
+ env_.clock().CurrentTime()
+ :
+ // "Now" using a system clock, relative to the UNIX epoch (Jan 1,
+ // 1970, UTC), in microseconds. The system clock could be modified
+ // and is not necessarily monotonically increasing.
+ Timestamp::Micros(rtc::TimeUTCMicros());
num_pending_partial_reports_ = 2;
partial_report_timestamp_us_ = cache_now_us;
@@ -1767,7 +1782,8 @@
for (const auto& report_block_data : voice_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
transport_id, report_block_data, cricket::MEDIA_TYPE_AUDIO,
- audio_outbound_rtps, *report));
+ audio_outbound_rtps, *report,
+ stats_timestamp_with_environment_clock_));
}
}
}
@@ -1861,7 +1877,8 @@
for (const auto& report_block_data : video_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
transport_id, report_block_data, cricket::MEDIA_TYPE_VIDEO,
- video_outbound_rtps, *report));
+ video_outbound_rtps, *report,
+ stats_timestamp_with_environment_clock_));
}
}
}
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 1beef4c..3020360 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -60,6 +60,7 @@
public:
static rtc::scoped_refptr<RTCStatsCollector> Create(
PeerConnectionInternal* pc,
+ const Environment& env,
int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);
// Gets a recent stats report. If there is a report cached that is still fresh
@@ -95,7 +96,9 @@
DataChannelInterface::DataState state);
protected:
- RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
+ RTCStatsCollector(PeerConnectionInternal* pc,
+ const Environment& env,
+ int64_t cache_lifetime_us);
~RTCStatsCollector();
struct CertificateStatsPair {
@@ -252,6 +255,8 @@
rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);
PeerConnectionInternal* const pc_;
+ const Environment env_;
+ const bool stats_timestamp_with_environment_clock_;
rtc::Thread* const signaling_thread_;
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 23c5e8f..cdbddf2 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -427,10 +427,12 @@
class RTCStatsCollectorWrapper {
public:
explicit RTCStatsCollectorWrapper(
- rtc::scoped_refptr<FakePeerConnectionForStats> pc)
+ rtc::scoped_refptr<FakePeerConnectionForStats> pc,
+ const Environment& env)
: pc_(pc),
stats_collector_(
RTCStatsCollector::Create(pc.get(),
+ env,
50 * rtc::kNumMicrosecsPerMillisec)) {}
rtc::scoped_refptr<RTCStatsCollector> stats_collector() {
@@ -662,7 +664,7 @@
public:
RTCStatsCollectorTest()
: pc_(rtc::make_ref_counted<FakePeerConnectionForStats>()),
- stats_(new RTCStatsCollectorWrapper(pc_)),
+ stats_(new RTCStatsCollectorWrapper(pc_, CreateEnvironment())),
data_channel_controller_(
new FakeDataChannelController(pc_->network_thread())) {}
@@ -3294,6 +3296,7 @@
TEST_P(RTCStatsCollectorTestWithParamKind,
RTCRemoteInboundRtpStreamStatsCollectedFromReportBlock) {
const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
+ const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
const uint8_t kFractionLost = 12;
const TimeDelta kRoundTripTimeSample1 = TimeDelta::Millis(1'234);
const TimeDelta kRoundTripTimeSample2 = TimeDelta::Seconds(13);
@@ -3311,7 +3314,8 @@
report_block.SetCumulativeLost(7);
report_block.SetFractionLost(kFractionLost);
ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
+ report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
+ kReportBlockTimestamp);
report_block_data.AddRoundTripTimeSample(kRoundTripTimeSample1);
// Only the last sample should be exposed as the
// `RTCRemoteInboundRtpStreamStats::round_trip_time`.
@@ -3361,13 +3365,15 @@
TEST_P(RTCStatsCollectorTestWithParamKind,
RTCRemoteInboundRtpStreamStatsRttMissingBeforeMeasurement) {
constexpr Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
+ const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
rtcp::ReportBlock report_block;
// The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
// `source_ssrc`, "SSRC of the RTP packet sender".
report_block.SetMediaSsrc(12);
ReportBlockData report_block_data; // AddRoundTripTimeSample() not called.
- report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
+ report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
+ kReportBlockTimestamp);
AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
std::nullopt);
@@ -3387,6 +3393,7 @@
TEST_P(RTCStatsCollectorTestWithParamKind,
RTCRemoteInboundRtpStreamStatsWithTimestampFromReportBlock) {
const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
+ const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
fake_clock_.SetTime(kReportBlockTimestampUtc);
rtcp::ReportBlock report_block;
@@ -3394,7 +3401,8 @@
// `source_ssrc`, "SSRC of the RTP packet sender".
report_block.SetMediaSsrc(12);
ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
+ report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
+ kReportBlockTimestamp);
AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
std::nullopt);
@@ -3419,6 +3427,7 @@
TEST_P(RTCStatsCollectorTestWithParamKind,
RTCRemoteInboundRtpStreamStatsWithCodecBasedMembers) {
const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
+ const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
fake_clock_.SetTime(kReportBlockTimestampUtc);
rtcp::ReportBlock report_block;
@@ -3427,7 +3436,8 @@
report_block.SetMediaSsrc(12);
report_block.SetJitter(5000);
ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
+ report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
+ kReportBlockTimestamp);
RtpCodecParameters codec;
codec.payload_type = 3;
@@ -3455,6 +3465,7 @@
TEST_P(RTCStatsCollectorTestWithParamKind,
RTCRemoteInboundRtpStreamStatsWithRtcpTransport) {
const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
+ const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
fake_clock_.SetTime(kReportBlockTimestampUtc);
rtcp::ReportBlock report_block;
@@ -3462,7 +3473,8 @@
// `source_ssrc`, "SSRC of the RTP packet sender".
report_block.SetMediaSsrc(12);
ReportBlockData report_block_data;
- report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
+ report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
+ kReportBlockTimestamp);
cricket::TransportChannelStats rtp_transport_channel_stats;
rtp_transport_channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
@@ -3747,9 +3759,10 @@
public:
static rtc::scoped_refptr<FakeRTCStatsCollector> Create(
PeerConnectionInternal* pc,
+ const Environment& env,
int64_t cache_lifetime_us) {
return rtc::scoped_refptr<FakeRTCStatsCollector>(
- new rtc::RefCountedObject<FakeRTCStatsCollector>(pc,
+ new rtc::RefCountedObject<FakeRTCStatsCollector>(pc, env,
cache_lifetime_us));
}
@@ -3794,8 +3807,10 @@
}
protected:
- FakeRTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime)
- : RTCStatsCollector(pc, cache_lifetime),
+ FakeRTCStatsCollector(PeerConnectionInternal* pc,
+ const Environment& env,
+ int64_t cache_lifetime)
+ : RTCStatsCollector(pc, env, cache_lifetime),
signaling_thread_(pc->signaling_thread()),
worker_thread_(pc->worker_thread()),
network_thread_(pc->network_thread()) {}
@@ -3845,7 +3860,7 @@
rtc::AutoThread main_thread_;
auto pc = rtc::make_ref_counted<FakePeerConnectionForStats>();
rtc::scoped_refptr<FakeRTCStatsCollector> stats_collector(
- FakeRTCStatsCollector::Create(pc.get(),
+ FakeRTCStatsCollector::Create(pc.get(), CreateEnvironment(),
50 * rtc::kNumMicrosecsPerMillisec));
stats_collector->VerifyThreadUsageAndResultsMerging();
}
diff --git a/video/send_statistics_proxy_unittest.cc b/video/send_statistics_proxy_unittest.cc
index 01b0349..0d022c7 100644
--- a/video/send_statistics_proxy_unittest.cc
+++ b/video/send_statistics_proxy_unittest.cc
@@ -199,7 +199,8 @@
report_block.SetFractionLost(offset + 2);
report_block.SetJitter(offset + 3);
ReportBlockData data;
- data.SetReportBlock(/*sender_ssrc=*/0, report_block, Timestamp::Zero());
+ data.SetReportBlock(/*sender_ssrc=*/0, report_block, Timestamp::Zero(),
+ Timestamp::Zero());
expected_.substreams[ssrc].report_block_data = data;
callback->OnReportBlockDataUpdated(data);
@@ -214,7 +215,8 @@
report_block.SetFractionLost(offset + 2);
report_block.SetJitter(offset + 3);
ReportBlockData data;
- data.SetReportBlock(/*sender_ssrc=*/0, report_block, Timestamp::Zero());
+ data.SetReportBlock(/*sender_ssrc=*/0, report_block, Timestamp::Zero(),
+ Timestamp::Zero());
expected_.substreams[ssrc].report_block_data = data;
callback->OnReportBlockDataUpdated(data);
@@ -2319,7 +2321,7 @@
rtcp::ReportBlock report_block;
report_block.SetMediaSsrc(excluded_ssrc);
ReportBlockData data;
- data.SetReportBlock(0, report_block, Timestamp::Zero());
+ data.SetReportBlock(0, report_block, Timestamp::Zero(), Timestamp::Zero());
rtcp_callback->OnReportBlockDataUpdated(data);
// From BitrateStatisticsObserver.
@@ -2372,7 +2374,7 @@
rtcp::ReportBlock report_block;
report_block.SetMediaSsrc(config_.rtp.ssrcs[0]);
ReportBlockData data;
- data.SetReportBlock(0, report_block, Timestamp::Zero());
+ data.SetReportBlock(0, report_block, Timestamp::Zero(), Timestamp::Zero());
rtcp_callback->OnReportBlockDataUpdated(data);
// Report stats for second SSRC to make sure it's not outdated along with the
