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Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.

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This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}
This commit is contained in:
Henrik Boström
2019-04-15 17:32:00 +02:00
committed by Commit Bot
parent 3d11e2f81c
commit 01738c63aa
12 changed files with 69 additions and 8 deletions
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20
pc/rtc_stats_collector_unittest.cc
View File

@ -1625,6 +1625,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
voice_media_info.receivers[0].codec_payload_type = 42;
voice_media_info.receivers[0].jitter_ms = 4500;
voice_media_info.receivers[0].fraction_lost = 5.5f;
voice_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
@ -1656,8 +1658,21 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
expected_audio.packets_received = 2;
expected_audio.bytes_received = 3;
expected_audio.packets_lost = -1;
// |expected_audio.last_packet_received_timestamp| should be undefined.
expected_audio.jitter = 4.5;
expected_audio.fraction_lost = 5.5;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(),
expected_audio);
// Set previously undefined values and "GetStats" again.
voice_media_info.receivers[0].last_packet_received_timestamp_ms = 3000;
expected_audio.last_packet_received_timestamp = 3.0;
voice_media_channel->SetStats(voice_media_info);
report = stats_->GetFreshStatsReport();
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(),
@ -1684,6 +1699,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
video_media_info.receivers[0].nacks_sent = 7;
video_media_info.receivers[0].frames_decoded = 8;
video_media_info.receivers[0].qp_sum = absl::nullopt;
video_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
video_media_info.receivers[0].content_type = VideoContentType::UNSPECIFIED;
RtpCodecParameters codec_parameters;
@ -1719,6 +1736,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
expected_video.fraction_lost = 4.5;
expected_video.frames_decoded = 8;
// |expected_video.qp_sum| should be undefined.
// |expected_video.last_packet_received_timestamp| should be undefined.
// |expected_video.content_type| should be undefined.
ASSERT_TRUE(report->Get(expected_video.id()));
@ -1728,7 +1746,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
// Set previously undefined values and "GetStats" again.
video_media_info.receivers[0].qp_sum = 9;
video_media_info.receivers[0].last_packet_received_timestamp_ms = 1000;
expected_video.qp_sum = 9;
expected_video.last_packet_received_timestamp = 1.0;
video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
expected_video.content_type = "screenshare";
video_media_channel->SetStats(video_media_info);