Fix unspecified time origin for `lastPacketReceivedTimestamp`
`RTCInboundRtpStreamStats.lastPacketReceivedTimestamp` must be a time
value in milliseconds with Unix epoch as time origin (see
bugs.webrtc.org/12605#c4).
This change fixes both audio and video `RTCInboundRtpStreamStats` stats.
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12605
Change-Id: I68157fcf01a5933f3d4e5d3918b4a9d3fbd64f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33547}
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 4c399a1..26c8cdd 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -24,9 +24,14 @@
#include "system_wrappers/include/clock.h"
namespace webrtc {
+namespace {
+constexpr int64_t kStatisticsTimeoutMs = 8000;
+constexpr int64_t kStatisticsProcessIntervalMs = 1000;
-const int64_t kStatisticsTimeoutMs = 8000;
-const int64_t kStatisticsProcessIntervalMs = 1000;
+// Number of seconds since 1900 January 1 00:00 GMT (see
+// https://tools.ietf.org/html/rfc868).
+constexpr int64_t kNtpJan1970Millisecs = 2'208'988'800'000;
+} // namespace
StreamStatistician::~StreamStatistician() {}
@@ -35,6 +40,9 @@
int max_reordering_threshold)
: ssrc_(ssrc),
clock_(clock),
+ delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() -
+ clock_->TimeInMilliseconds() -
+ kNtpJan1970Millisecs),
incoming_bitrate_(kStatisticsProcessIntervalMs,
RateStatistics::kBpsScale),
max_reordering_threshold_(max_reordering_threshold),
@@ -172,8 +180,11 @@
// TODO(nisse): Can we return a float instead?
// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
stats.jitter = jitter_q4_ >> 4;
- stats.last_packet_received_timestamp_ms =
- receive_counters_.last_packet_received_timestamp_ms;
+ if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
+ stats.last_packet_received_timestamp_ms =
+ *receive_counters_.last_packet_received_timestamp_ms +
+ delta_internal_unix_epoch_ms_;
+ }
stats.packet_counter = receive_counters_.transmitted;
return stats;
}
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.h b/modules/rtp_rtcp/source/receive_statistics_impl.h
index 2456f93..be56f4b 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -73,6 +73,8 @@
const uint32_t ssrc_;
Clock* const clock_;
+ // Delta used to map internal timestamps to Unix epoch ones.
+ const int64_t delta_internal_unix_epoch_ms_;
RateStatistics incoming_bitrate_;
// In number of packets or sequence numbers.
int max_reordering_threshold_;
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 36ee542..4c9dfa4 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -372,6 +372,7 @@
*voice_receiver_info.last_packet_received_timestamp_ms);
}
if (voice_receiver_info.estimated_playout_ntp_timestamp_ms) {
+ // TODO(bugs.webrtc.org/10529): Fix time origin.
inbound_audio->estimated_playout_timestamp = static_cast<double>(
*voice_receiver_info.estimated_playout_ntp_timestamp_ms);
}
@@ -471,17 +472,16 @@
inbound_video->total_squared_inter_frame_delay =
video_receiver_info.total_squared_inter_frame_delay;
if (video_receiver_info.last_packet_received_timestamp_ms) {
- inbound_video->last_packet_received_timestamp =
- static_cast<double>(
- *video_receiver_info.last_packet_received_timestamp_ms) /
- rtc::kNumMillisecsPerSec;
+ inbound_video->last_packet_received_timestamp = static_cast<double>(
+ *video_receiver_info.last_packet_received_timestamp_ms);
}
if (video_receiver_info.estimated_playout_ntp_timestamp_ms) {
+ // TODO(bugs.webrtc.org/10529): Fix time origin if needed.
inbound_video->estimated_playout_timestamp = static_cast<double>(
*video_receiver_info.estimated_playout_ntp_timestamp_ms);
}
- // TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
- // optional, support the "unspecified" value.
+ // TODO(bugs.webrtc.org/10529): When info's |content_info| is optional
+ // support the "unspecified" value.
if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)
inbound_video->content_type = RTCContentType::kScreenshare;
if (!video_receiver_info.decoder_implementation_name.empty()) {
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 897226d..655f7e6 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2107,7 +2107,7 @@
video_media_info.receivers[0].qp_sum = 9;
expected_video.qp_sum = 9;
video_media_info.receivers[0].last_packet_received_timestamp_ms = 1000;
- expected_video.last_packet_received_timestamp = 1.0;
+ expected_video.last_packet_received_timestamp = 1000.0;
video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
expected_video.content_type = "screenshare";
video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 1234;
