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Propagate time of the last received packet with Timestamp type

Browse source 
Bug: webrtc:13757
Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40211}
This commit is contained in:
Danil Chapovalov 2023-06-02 14:54:45 +02:00 committed by WebRTC LUCI CQ
parent 9a34d80fc4
commit 54e95bc562
13 changed files with 29 additions and 31 deletions
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3
audio/audio_receive_stream.cc
View file

@ -266,8 +266,7 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
stats.packets_lost = call_stats.cumulativeLost;
stats.nacks_sent = call_stats.nacks_sent;
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
stats.last_packet_received_timestamp_ms =
call_stats.last_packet_received_timestamp_ms;
stats.last_packet_received = call_stats.last_packet_received;
stats.codec_name = receive_codec->second.name;
stats.codec_payload_type = receive_codec->first;
int clockrate_khz = receive_codec->second.clockrate_hz / 1000;

5
audio/channel_receive.cc
View file

@ -828,13 +828,12 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
rtp_stats.packet_counter.header_bytes +
rtp_stats.packet_counter.padding_bytes;
stats.packetsReceived = rtp_stats.packet_counter.packets;
stats.last_packet_received_timestamp_ms =
rtp_stats.last_packet_received_timestamp_ms;
stats.last_packet_received = rtp_stats.last_packet_received;
} else {
stats.payload_bytes_received = 0;
stats.header_and_padding_bytes_received = 0;
stats.packetsReceived = 0;
stats.last_packet_received_timestamp_ms = absl::nullopt;
stats.last_packet_received = absl::nullopt;
}
{

2
audio/channel_receive.h
View file

@ -63,7 +63,7 @@ struct CallReceiveStatistics {
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
absl::optional<int64_t> last_packet_received_timestamp_ms;
absl::optional<Timestamp> last_packet_received;
// Remote outbound stats derived by the received RTCP sender reports.
// Note that the timestamps below correspond to the time elapsed since the
// Unix epoch.

2
call/audio_receive_stream.h
View file

@ -83,7 +83,7 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
absl::optional<int64_t> last_packet_received_timestamp_ms;
absl::optional<Timestamp> last_packet_received;
uint64_t jitter_buffer_flushes = 0;
double relative_packet_arrival_delay_seconds = 0.0;
int32_t interruption_count = 0;

1
media/BUILD.gn
View file

@ -250,6 +250,7 @@ rtc_source_set("media_channel") {
"../api/transport:datagram_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../api/video:video_rtp_headers",

3
media/base/media_channel.h
View file

@ -33,6 +33,7 @@
#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_content_type.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
@ -458,7 +459,7 @@ struct MediaReceiverInfo {
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
absl::optional<int64_t> last_packet_received_timestamp_ms;
absl::optional<webrtc::Timestamp> last_packet_received;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
std::string codec_name;

3
media/engine/webrtc_video_engine.cc
View file

@ -3572,8 +3572,7 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
info.total_assembly_time = stats.total_assembly_time;
info.frames_assembled_from_multiple_packets =
stats.frames_assembled_from_multiple_packets;
info.last_packet_received_timestamp_ms =
stats.rtp_stats.last_packet_received_timestamp_ms;
info.last_packet_received = stats.rtp_stats.last_packet_received;
info.estimated_playout_ntp_timestamp_ms =
stats.estimated_playout_ntp_timestamp_ms;
info.first_frame_received_to_decoded_ms =

3
media/engine/webrtc_voice_engine.cc
View file

@ -2496,8 +2496,7 @@ bool WebRtcVoiceMediaChannel::GetReceiveStats(VoiceMediaReceiveInfo* info,
rinfo.decoding_plc_cng = stats.decoding_plc_cng;
rinfo.decoding_muted_output = stats.decoding_muted_output;
rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
rinfo.last_packet_received_timestamp_ms =
stats.last_packet_received_timestamp_ms;
rinfo.last_packet_received = stats.last_packet_received;
rinfo.estimated_playout_ntp_timestamp_ms =
stats.estimated_playout_ntp_timestamp_ms;
rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes;

7
modules/rtp_rtcp/include/rtp_rtcp_defines.h
View file

@ -387,9 +387,12 @@ struct RtpReceiveStats {
// Interarrival jitter in time.
webrtc::TimeDelta interarrival_jitter = webrtc::TimeDelta::Zero();
// Timestamp and counters exposed in RTCInboundRtpStreamStats, see
// Time of the last packet received in unix epoch,
// i.e. Timestamp::Zero() represents 1st Jan 1970 00:00
absl::optional<Timestamp> last_packet_received;
// Counters exposed in RTCInboundRtpStreamStats, see
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
absl::optional<int64_t> last_packet_received_timestamp_ms;
RtpPacketCounter packet_counter;
};

4
modules/rtp_rtcp/source/receive_statistics_impl.cc
View file

@ -227,8 +227,8 @@ RtpReceiveStats StreamStatisticianImpl::GetStats() const {
TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_;
}
if (last_receive_time_.has_value()) {
stats.last_packet_received_timestamp_ms =
(*last_receive_time_ + delta_internal_unix_epoch_).ms();
stats.last_packet_received =
*last_receive_time_ + delta_internal_unix_epoch_;
}
stats.packet_counter = receive_counters_.transmitted;
return stats;

4
modules/rtp_rtcp/source/receive_statistics_unittest.cc
View file

@ -595,13 +595,13 @@ TEST_P(ReceiveStatisticsTest, LastPacketReceivedTimestamp) {
RtpReceiveStats counters =
receive_statistics_->GetStatistician(kSsrc1)->GetStats();
EXPECT_EQ(42, counters.last_packet_received_timestamp_ms);
EXPECT_EQ(counters.last_packet_received, Timestamp::Millis(42));
clock_.AdvanceTimeMilliseconds(3);
packet1_.SetSequenceNumber(101);
receive_statistics_->OnRtpPacket(packet1_);
counters = receive_statistics_->GetStatistician(kSsrc1)->GetStats();
EXPECT_EQ(45, counters.last_packet_received_timestamp_ms);
EXPECT_EQ(counters.last_packet_received, Timestamp::Millis(45));
}
TEST_P(ReceiveStatisticsTest, SimpleJitterComputation) {

12
pc/rtc_stats_collector.cc
View file

@ -474,9 +474,9 @@ std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats(
voice_receiver_info.total_output_duration;
// `fir_count` and `pli_count` are only valid for video and are
// purposefully left undefined for audio.
if (voice_receiver_info.last_packet_received_timestamp_ms.has_value()) {
inbound_audio->last_packet_received_timestamp = static_cast<double>(
*voice_receiver_info.last_packet_received_timestamp_ms);
if (voice_receiver_info.last_packet_received.has_value()) {
inbound_audio->last_packet_received_timestamp =
voice_receiver_info.last_packet_received->ms<double>();
}
if (voice_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) {
// TODO(bugs.webrtc.org/10529): Fix time origin.
@ -649,9 +649,9 @@ CreateInboundRTPStreamStatsFromVideoReceiverInfo(
inbound_video->min_playout_delay =
static_cast<double>(video_receiver_info.min_playout_delay_ms) /
rtc::kNumMillisecsPerSec;
if (video_receiver_info.last_packet_received_timestamp_ms.has_value()) {
inbound_video->last_packet_received_timestamp = static_cast<double>(
*video_receiver_info.last_packet_received_timestamp_ms);
if (video_receiver_info.last_packet_received.has_value()) {
inbound_video->last_packet_received_timestamp =
video_receiver_info.last_packet_received->ms<double>();
}
if (video_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) {
// TODO(bugs.webrtc.org/10529): Fix time origin if needed.

11
pc/rtc_stats_collector_unittest.cc
View file

@ -2183,9 +2183,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
voice_media_info.receivers[0].relative_packet_arrival_delay_seconds = 16;
voice_media_info.receivers[0].interruption_count = 7788;
voice_media_info.receivers[0].total_interruption_duration_ms = 778899;
voice_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
voice_media_info.receivers[0].last_packet_received = absl::nullopt;
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
@ -2251,7 +2249,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
expected_audio);
// Set previously undefined values and "GetStats" again.
voice_media_info.receivers[0].last_packet_received_timestamp_ms = 3000;
voice_media_info.receivers[0].last_packet_received = Timestamp::Seconds(3);
expected_audio.last_packet_received_timestamp = 3000.0;
voice_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 4567;
expected_audio.estimated_playout_timestamp = 4567;
@ -2344,8 +2342,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
video_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
video_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
video_media_info.receivers[0].jitter_buffer_emitted_count = 13;
video_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
video_media_info.receivers[0].last_packet_received = absl::nullopt;
video_media_info.receivers[0].content_type = VideoContentType::UNSPECIFIED;
video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms =
absl::nullopt;
@ -2427,7 +2424,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
// Set previously undefined values and "GetStats" again.
video_media_info.receivers[0].qp_sum = 9;
expected_video.qp_sum = 9;
video_media_info.receivers[0].last_packet_received_timestamp_ms = 1000;
video_media_info.receivers[0].last_packet_received = Timestamp::Seconds(1);
expected_video.last_packet_received_timestamp = 1000.0;
video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
expected_video.content_type = "screenshare";