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Add jitterBufferDelay and jitterBufferEmittedCount stats for video

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jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.

Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
This commit is contained in:
Byoungchan Lee 2021-06-29 22:09:18 +09:00 committed by WebRTC LUCI CQ
parent 7d0b422af6
commit 899b29eb25
4 changed files with 20 additions and 21 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

14
media/base/media_channel.h
View File

@ -426,6 +426,12 @@ struct MediaReceiverInfo {
int64_t header_and_padding_bytes_rcvd = 0;
int packets_rcvd = 0;
int packets_lost = 0;
// Jitter (network-related) latency (cumulative).
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
double jitter_buffer_delay_seconds = 0.0;
// Number of observations for cumulative jitter latency.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferemittedcount
uint64_t jitter_buffer_emitted_count = 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
@ -469,8 +475,6 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
uint64_t concealed_samples = 0;
uint64_t silent_concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
double jitter_buffer_target_delay_seconds = 0.0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
@ -616,12 +620,6 @@ struct VideoReceiverInfo : public MediaReceiverInfo {
int max_decode_ms = 0;
// Jitter (network-related) latency.
int jitter_buffer_ms = 0;
// Jitter (network-related) latency (cumulative).
// https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferdelay
double jitter_buffer_delay_seconds = 0;
// Number of observations for cumulative jitter latency.
// https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferemittedcount
uint64_t jitter_buffer_emitted_count = 0;
// Requested minimum playout latency.
int min_playout_delay_ms = 0;
// Requested latency to account for rendering delay.

8
pc/rtc_stats_collector.cc
View File

@ -333,6 +333,10 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
static_cast<uint64_t>(media_receiver_info.header_and_padding_bytes_rcvd);
inbound_stats->packets_lost =
static_cast<int32_t>(media_receiver_info.packets_lost);
inbound_stats->jitter_buffer_delay =
media_receiver_info.jitter_buffer_delay_seconds;
inbound_stats->jitter_buffer_emitted_count =
media_receiver_info.jitter_buffer_emitted_count;
}
std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
@ -353,10 +357,6 @@ std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
}
inbound_audio->jitter = static_cast<double>(voice_receiver_info.jitter_ms) /
rtc::kNumMillisecsPerSec;
inbound_audio->jitter_buffer_delay =
voice_receiver_info.jitter_buffer_delay_seconds;
inbound_audio->jitter_buffer_emitted_count =
voice_receiver_info.jitter_buffer_emitted_count;
inbound_audio->total_samples_received =
voice_receiver_info.total_samples_received;
inbound_audio->concealed_samples = voice_receiver_info.concealed_samples;

8
pc/rtc_stats_collector_unittest.cc
View File

@ -1737,7 +1737,7 @@ TEST_F(RTCStatsCollectorTest,
voice_receiver_info.inserted_samples_for_deceleration = 987;
voice_receiver_info.removed_samples_for_acceleration = 876;
voice_receiver_info.silent_concealed_samples = 765;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
voice_receiver_info.jitter_buffer_delay_seconds = 3.456;
voice_receiver_info.jitter_buffer_emitted_count = 13;
voice_receiver_info.jitter_buffer_target_delay_seconds = 7.894;
voice_receiver_info.jitter_buffer_flushes = 7;
@ -1782,7 +1782,7 @@ TEST_F(RTCStatsCollectorTest,
expected_remote_audio_track.inserted_samples_for_deceleration = 987;
expected_remote_audio_track.removed_samples_for_acceleration = 876;
expected_remote_audio_track.silent_concealed_samples = 765;
expected_remote_audio_track.jitter_buffer_delay = 3456;
expected_remote_audio_track.jitter_buffer_delay = 3.456;
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
expected_remote_audio_track.jitter_buffer_target_delay = 7.894;
expected_remote_audio_track.jitter_buffer_flushes = 7;
@ -2068,6 +2068,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
video_media_info.receivers[0].total_inter_frame_delay = 0.123;
video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456;
video_media_info.receivers[0].jitter_ms = 1199;
video_media_info.receivers[0].jitter_buffer_delay_seconds = 3.456;
video_media_info.receivers[0].jitter_buffer_emitted_count = 13;
video_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
@ -2115,6 +2117,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
expected_video.total_inter_frame_delay = 0.123;
expected_video.total_squared_inter_frame_delay = 0.00456;
expected_video.jitter = 1.199;
expected_video.jitter_buffer_delay = 3.456;
expected_video.jitter_buffer_emitted_count = 13;
// |expected_video.last_packet_received_timestamp| should be undefined.
// |expected_video.content_type| should be undefined.
// |expected_video.decoder_implementation| should be undefined.

11
pc/rtc_stats_integrationtest.cc
View File

@ -840,11 +840,12 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsUndefined(inbound_stream.frames_per_second);
}
verifier.TestMemberIsUndefined(inbound_stream.frame_bit_depth);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.jitter_buffer_emitted_count);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsUndefined(inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
inbound_stream.jitter_buffer_emitted_count);
verifier.TestMemberIsUndefined(inbound_stream.total_samples_received);
verifier.TestMemberIsUndefined(inbound_stream.concealed_samples);
verifier.TestMemberIsUndefined(inbound_stream.silent_concealed_samples);
@ -864,10 +865,6 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsUndefined(inbound_stream.fir_count);
verifier.TestMemberIsUndefined(inbound_stream.pli_count);
verifier.TestMemberIsUndefined(inbound_stream.nack_count);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.jitter_buffer_emitted_count);
verifier.TestMemberIsPositive<uint64_t>(
inbound_stream.total_samples_received);
verifier.TestMemberIsNonNegative<uint64_t>(