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Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

1
api/stats/rtcstats_objects.h
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@ -397,6 +397,7 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;
// TODO(hbos): Collect and populate this value for both "audio" and "video",
// currently not collected for "video". https://bugs.webrtc.org/7065
RTCStatsMember<double> jitter;

2
audio/audio_receive_stream.cc
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@ -190,6 +190,8 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
stats.packets_lost = call_stats.cumulativeLost;
stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
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.codec_name = receive_codec->second.name;
stats.codec_payload_type = receive_codec->first;
stats.ext_seqnum = call_stats.extendedMax;

22
audio/channel_receive.cc
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@ -757,17 +757,23 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
stats.rttMs = GetRTT();
// --- Data counters
size_t bytesReceived(0);
uint32_t packetsReceived(0);
if (statistician) {
statistician->GetDataCounters(&bytesReceived, &packetsReceived);
StreamDataCounters data_counters;
statistician->GetReceiveStreamDataCounters(&data_counters);
// TODO(http://crbug.com/webrtc/10525): Bytes received should only include
// payload bytes, not header and padding bytes.
stats.bytesReceived = data_counters.transmitted.payload_bytes +
data_counters.transmitted.header_bytes +
data_counters.transmitted.padding_bytes;
stats.packetsReceived = data_counters.transmitted.packets;
stats.last_packet_received_timestamp_ms =
data_counters.last_packet_received_timestamp_ms;
} else {
stats.bytesReceived = 0;
stats.packetsReceived = 0;
stats.last_packet_received_timestamp_ms = absl::nullopt;
}
stats.bytesReceived = bytesReceived;
stats.packetsReceived = packetsReceived;
// --- Timestamps
{
rtc::CritScope lock(&ts_stats_lock_);

4
audio/channel_receive.h
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@ -59,6 +59,10 @@ struct CallReceiveStatistics {
// The capture ntp time (in local timebase) of the first played out audio
// frame.
int64_t capture_start_ntp_time_ms_;
// 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;
};
namespace voe {

4
call/audio_receive_stream.h
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@ -73,6 +73,10 @@ class AudioReceiveStream {
int32_t decoding_plc_cng = 0;
int32_t decoding_muted_output = 0;
int64_t capture_start_ntp_time_ms = 0;
// 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;
uint64_t jitter_buffer_flushes = 0;
double relative_packet_arrival_delay_seconds = 0.0;
};

4
media/base/media_channel.h
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@ -435,6 +435,10 @@ struct MediaReceiverInfo {
int packets_rcvd = 0;
int packets_lost = 0;
float fraction_lost = 0.0f;
// 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;
std::string codec_name;
absl::optional<int> codec_payload_type;
std::vector<SsrcReceiverInfo> local_stats;

2
media/engine/webrtc_video_engine.cc
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@ -2697,6 +2697,8 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
info.frames_decoded = stats.frames_decoded;
info.frames_rendered = stats.frames_rendered;
info.qp_sum = stats.qp_sum;
info.last_packet_received_timestamp_ms =
stats.rtp_stats.last_packet_received_timestamp_ms;
info.first_frame_received_to_decoded_ms =
stats.first_frame_received_to_decoded_ms;
info.interframe_delay_max_ms = stats.interframe_delay_max_ms;

2
media/engine/webrtc_voice_engine.cc
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@ -2267,6 +2267,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* 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.jitter_buffer_flushes = stats.jitter_buffer_flushes;
rinfo.relative_packet_arrival_delay_seconds =
stats.relative_packet_arrival_delay_seconds;

12
pc/rtc_stats_collector.cc
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@ -244,6 +244,12 @@ void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
rtc::kNumMillisecsPerSec;
// |fir_count|, |pli_count| and |sli_count| are only valid for video and are
// purposefully left undefined for audio.
if (voice_receiver_info.last_packet_received_timestamp_ms) {
inbound_audio->last_packet_received_timestamp =
static_cast<double>(
*voice_receiver_info.last_packet_received_timestamp_ms) /
rtc::kNumMillisecsPerSec;
}
}
void SetInboundRTPStreamStatsFromVideoReceiverInfo(
@ -267,6 +273,12 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
inbound_video->frames_decoded = video_receiver_info.frames_decoded;
if (video_receiver_info.qp_sum)
inbound_video->qp_sum = *video_receiver_info.qp_sum;
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;
}
// TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
// optional, support the "unspecified" value.
if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)

20
pc/rtc_stats_collector_unittest.cc
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@ -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);

1
pc/rtc_stats_integrationtest.cc
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@ -719,6 +719,7 @@ class RTCStatsReportVerifier {
// packets_lost is defined as signed, but this should never happen in
// this test. See RFC 3550.
verifier.TestMemberIsNonNegative<int32_t>(inbound_stream.packets_lost);
verifier.TestMemberIsDefined(inbound_stream.last_packet_received_timestamp);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsUndefined(inbound_stream.jitter);

3
stats/rtcstats_objects.cc
View File

@ -578,6 +578,7 @@ WEBRTC_RTCSTATS_IMPL(
&packets_received,
&bytes_received,
&packets_lost,
&last_packet_received_timestamp,
&jitter,
&fraction_lost,
&round_trip_time,
@ -605,6 +606,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
packets_received("packetsReceived"),
bytes_received("bytesReceived"),
packets_lost("packetsLost"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
jitter("jitter"),
fraction_lost("fractionLost"),
round_trip_time("roundTripTime"),
@ -627,6 +629,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
packets_received(other.packets_received),
bytes_received(other.bytes_received),
packets_lost(other.packets_lost),
last_packet_received_timestamp(other.last_packet_received_timestamp),
jitter(other.jitter),
fraction_lost(other.fraction_lost),
round_trip_time(other.round_trip_time),