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Fix unspecified time origin for lastPacketReceivedTimestamp

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`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}
This commit is contained in:
Alessio Bazzica
2021-03-24 08:51:26 +01:00
committed by Commit Bot
parent 9054aa8904
commit 5cf8c2c501
4 changed files with 24 additions and 11 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -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 @@ StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc,
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 @@ RtpReceiveStats StreamStatisticianImpl::GetStats() const {
// 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;
if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
stats.last_packet_received_timestamp_ms =
receive_counters_.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;
}

2
modules/rtp_rtcp/source/receive_statistics_impl.h
View File

@ -73,6 +73,8 @@ class StreamStatisticianImpl : public StreamStatisticianImplInterface {
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_;

12
pc/rtc_stats_collector.cc
View File

@ -372,6 +372,7 @@ std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
*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 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
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()) {

2
pc/rtc_stats_collector_unittest.cc
View File

@ -2107,7 +2107,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
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;