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Change timestamps type from int64 to Timestamp in MediaReceiverInfo.

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We should use the Timestamp type, rather then int64, to store timestamps. In https://webrtc-review.googlesource.com/c/src/+/365001/ an additional int64 timestamp was added (last_sender_report_timestamp_ms).

This CL fixes the new timestamp, as well as other similar timestamps in MediaReceiverInfo (last_sender_report_utc_timestamp_ms and last_sender_report_remote_utc_timestamp_ms).

Bug: webrtc:372393493
Change-Id: I0e473730e85a69ec595b421e2c3db920364008eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43248}
This commit is contained in:
Olov Brändström 2024-10-15 15:10:52 +02:00 committed by WebRTC LUCI CQ
parent 9c21f6386f
commit 558c2dc539
11 changed files with 58 additions and 65 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

11
audio/audio_receive_stream.cc
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@ -332,12 +332,11 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
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_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.last_sender_report_timestamp = call_stats.last_sender_report_timestamp;
stats.last_sender_report_utc_timestamp =
call_stats.last_sender_report_utc_timestamp;
stats.last_sender_report_remote_utc_timestamp =
call_stats.last_sender_report_remote_utc_timestamp;
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;

15
audio/channel_receive.cc
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@ -872,14 +872,13 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
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.ms<int64_t>();
stats.last_sender_report_utc_timestamp_ms =
rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
stats.last_sender_report_remote_utc_timestamp_ms =
rtcp_sr_stats->last_remote_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
stats.last_sender_report_timestamp = rtcp_sr_stats->last_arrival_timestamp;
stats.last_sender_report_utc_timestamp =
Timestamp::Millis(rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs);
stats.last_sender_report_remote_utc_timestamp =
Timestamp::Millis(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;

8
audio/channel_receive.h
View File

@ -67,13 +67,11 @@ struct CallReceiveStatistics {
// Note that the timestamps below correspond to the time elapsed since the
// Unix epoch.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
// TODO: bugs.webrtc.org/372393493 - timestamps should use the type Timestamp,
// not int64_t.
std::optional<int64_t> last_sender_report_timestamp_ms;
std::optional<Timestamp> last_sender_report_timestamp;
// TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
// issue is fixed.
std::optional<int64_t> last_sender_report_utc_timestamp_ms;
std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
std::optional<Timestamp> last_sender_report_utc_timestamp;
std::optional<Timestamp> last_sender_report_remote_utc_timestamp;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;

6
call/audio_receive_stream.h
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@ -100,11 +100,11 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
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<Timestamp> last_sender_report_timestamp;
// TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
// issue is fixed.
std::optional<int64_t> last_sender_report_utc_timestamp_ms;
std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
std::optional<Timestamp> last_sender_report_utc_timestamp;
std::optional<Timestamp> last_sender_report_remote_utc_timestamp;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;

6
call/video_receive_stream.h
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@ -175,11 +175,11 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// 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<Timestamp> last_sender_report_timestamp;
// TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
// issue is fixed.
std::optional<int64_t> last_sender_report_utc_timestamp_ms;
std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
std::optional<Timestamp> last_sender_report_utc_timestamp;
std::optional<Timestamp> last_sender_report_remote_utc_timestamp;
uint32_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;

8
media/base/media_channel.h
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@ -482,13 +482,11 @@ struct MediaReceiverInfo {
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
// TODO: bugs.webrtc.org/372393493 - timestamps should use the type Timestamp,
// not int64_t.
std::optional<int64_t> last_sender_report_timestamp_ms;
std::optional<webrtc::Timestamp> last_sender_report_timestamp;
// TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
// issue is fixed.
std::optional<int64_t> last_sender_report_utc_timestamp_ms;
std::optional<int64_t> last_sender_report_remote_utc_timestamp_ms;
std::optional<webrtc::Timestamp> last_sender_report_utc_timestamp;
std::optional<webrtc::Timestamp> last_sender_report_remote_utc_timestamp;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;

10
media/engine/webrtc_video_engine.cc
View File

@ -3845,11 +3845,11 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
}
// remote-outbound-rtp stats.
info.last_sender_report_timestamp_ms = stats.last_sender_report_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.last_sender_report_timestamp = stats.last_sender_report_timestamp;
info.last_sender_report_utc_timestamp =
stats.last_sender_report_utc_timestamp;
info.last_sender_report_remote_utc_timestamp =
stats.last_sender_report_remote_utc_timestamp;
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;

11
media/engine/webrtc_voice_engine.cc
View File

@ -2697,12 +2697,11 @@ bool WebRtcVoiceReceiveChannel::GetStats(VoiceMediaReceiveInfo* info,
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_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.last_sender_report_timestamp = stats.last_sender_report_timestamp;
rinfo.last_sender_report_utc_timestamp =
stats.last_sender_report_utc_timestamp;
rinfo.last_sender_report_remote_utc_timestamp =
stats.last_sender_report_remote_utc_timestamp;
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;

18
pc/rtc_stats_collector.cc
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@ -523,11 +523,11 @@ CreateRemoteOutboundMediaStreamStats(
const RTCInboundRtpStreamStats& inbound_audio_stats,
const std::string& transport_id,
const bool stats_timestamp_with_environment_clock) {
std::optional<int64_t> last_sender_report_timestamp_ms =
std::optional<Timestamp> last_sender_report_timestamp =
stats_timestamp_with_environment_clock
? media_receiver_info.last_sender_report_timestamp_ms
: media_receiver_info.last_sender_report_utc_timestamp_ms;
if (!last_sender_report_timestamp_ms.has_value()) {
? media_receiver_info.last_sender_report_timestamp
: media_receiver_info.last_sender_report_utc_timestamp;
if (!last_sender_report_timestamp.has_value()) {
// Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival
// timestamp is not available - i.e., until the first sender report is
// received.
@ -539,7 +539,7 @@ CreateRemoteOutboundMediaStreamStats(
auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>(
/*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
media_type, media_receiver_info.ssrc()),
Timestamp::Millis(*last_sender_report_timestamp_ms));
*last_sender_report_timestamp);
// Populate.
// - RTCRtpStreamStats.
@ -556,10 +556,10 @@ CreateRemoteOutboundMediaStreamStats(
stats->local_id = inbound_audio_stats.id();
// 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_utc_timestamp_ms);
RTC_DCHECK(
media_receiver_info.last_sender_report_remote_utc_timestamp.has_value());
stats->remote_timestamp =
media_receiver_info.last_sender_report_remote_utc_timestamp->ms<double>();
stats->reports_sent = media_receiver_info.sender_reports_reports_count;
if (media_receiver_info.round_trip_time.has_value()) {
stats->round_trip_time =

18
pc/rtc_stats_collector_unittest.cc
View File

@ -145,8 +145,9 @@ const int64_t kGetStatsReportTimeoutMs = 1000;
// Fake data used by `SetupExampleStatsVoiceGraph()` to fill in remote outbound
// stats.
constexpr int64_t kRemoteOutboundStatsTimestampMs = 123;
constexpr int64_t kRemoteOutboundStatsRemoteTimestampMs = 456;
constexpr Timestamp kRemoteOutboundStatsTimestamp = Timestamp::Millis(123);
constexpr Timestamp kRemoteOutboundStatsRemoteTimestamp =
Timestamp::Millis(456);
constexpr uint32_t kRemoteOutboundStatsPacketsSent = 7u;
constexpr uint64_t kRemoteOutboundStatsBytesSent = 8u;
constexpr uint64_t kRemoteOutboundStatsReportsCount = 9u;
@ -844,10 +845,10 @@ class RTCStatsCollectorTest : public ::testing::Test {
// remote-outbound-rtp
if (add_remote_outbound_stats) {
graph.remote_outbound_rtp_id = "ROA4";
media_info.receivers[0].last_sender_report_utc_timestamp_ms =
kRemoteOutboundStatsTimestampMs;
media_info.receivers[0].last_sender_report_remote_utc_timestamp_ms =
kRemoteOutboundStatsRemoteTimestampMs;
media_info.receivers[0].last_sender_report_utc_timestamp =
kRemoteOutboundStatsTimestamp;
media_info.receivers[0].last_sender_report_remote_utc_timestamp =
kRemoteOutboundStatsRemoteTimestamp;
media_info.receivers[0].sender_reports_packets_sent =
kRemoteOutboundStatsPacketsSent;
media_info.receivers[0].sender_reports_bytes_sent =
@ -3533,10 +3534,9 @@ TEST_F(RTCStatsCollectorTest, RTCRemoteOutboundRtpAudioStreamStatsCollected) {
const auto& remote_outbound_rtp =
graph.full_report->Get(graph.remote_outbound_rtp_id)
->cast_to<RTCRemoteOutboundRtpStreamStats>();
EXPECT_EQ(remote_outbound_rtp.timestamp(),
Timestamp::Millis(kRemoteOutboundStatsTimestampMs));
EXPECT_EQ(remote_outbound_rtp.timestamp(), kRemoteOutboundStatsTimestamp);
EXPECT_FLOAT_EQ(*remote_outbound_rtp.remote_timestamp,
static_cast<double>(kRemoteOutboundStatsRemoteTimestampMs));
kRemoteOutboundStatsRemoteTimestamp.ms<double>());
EXPECT_EQ(*remote_outbound_rtp.packets_sent, kRemoteOutboundStatsPacketsSent);
EXPECT_EQ(*remote_outbound_rtp.bytes_sent, kRemoteOutboundStatsBytesSent);
EXPECT_EQ(*remote_outbound_rtp.reports_sent,

12
video/video_receive_stream2.cc
View File

@ -572,12 +572,12 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
std::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_video_stream_receiver_.GetSenderReportStats();
if (rtcp_sr_stats) {
stats.last_sender_report_utc_timestamp_ms =
rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
stats.last_sender_report_remote_utc_timestamp_ms =
rtcp_sr_stats->last_remote_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs;
stats.last_sender_report_utc_timestamp =
Timestamp::Millis(rtcp_sr_stats->last_arrival_ntp_timestamp.ToMs() -
rtc::kNtpJan1970Millisecs);
stats.last_sender_report_remote_utc_timestamp =
Timestamp::Millis(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;