Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay):
- "WebRTC.Video.OnewayDelayInMs"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1351403008
Cr-Commit-Position: refs/heads/master@{#10180}
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index b6063a8..534428b 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -53,6 +53,10 @@
int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
if (decode_ms != -1)
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
+
+ int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
+ if (delay_ms != -1)
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
@@ -78,7 +82,8 @@
int target_delay_ms,
int jitter_buffer_ms,
int min_playout_delay_ms,
- int render_delay_ms) {
+ int render_delay_ms,
+ int64_t rtt_ms) {
rtc::CritScope lock(&crit_);
stats_.decode_ms = decode_ms;
stats_.max_decode_ms = max_decode_ms;
@@ -88,6 +93,9 @@
stats_.min_playout_delay_ms = min_playout_delay_ms;
stats_.render_delay_ms = render_delay_ms;
decode_time_counter_.Add(decode_ms);
+ // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
+ // render delay).
+ delay_counter_.Add(target_delay_ms + rtt_ms / 2);
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
diff --git a/webrtc/video/receive_statistics_proxy.h b/webrtc/video/receive_statistics_proxy.h
index 4f2550a..7bfa6d6 100644
--- a/webrtc/video/receive_statistics_proxy.h
+++ b/webrtc/video/receive_statistics_proxy.h
@@ -51,7 +51,8 @@
int target_delay_ms,
int jitter_buffer_ms,
int min_playout_delay_ms,
- int render_delay_ms);
+ int render_delay_ms,
+ int64_t rtt_ms);
// Overrides VCMReceiveStatisticsCallback.
void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) override;
@@ -94,6 +95,7 @@
SampleCounter render_width_counter_ GUARDED_BY(crit_);
SampleCounter render_height_counter_ GUARDED_BY(crit_);
SampleCounter decode_time_counter_ GUARDED_BY(crit_);
+ SampleCounter delay_counter_ GUARDED_BY(crit_);
ReportBlockStats report_block_stats_ GUARDED_BY(crit_);
};
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 7664fb9..264964d 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -115,6 +115,7 @@
report_block_stats_sender_(new ReportBlockStats()),
time_of_first_rtt_ms_(-1),
rtt_sum_ms_(0),
+ last_rtt_ms_(0),
num_rtts_(0),
rtp_rtcp_modules_(
CreateRtpRtcpModules(!sender,
@@ -1068,7 +1069,7 @@
return;
receive_stats_callback_->OnDecoderTiming(
decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
- jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
+ jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_);
}
int32_t ViEChannel::RequestKeyFrame() {
@@ -1102,6 +1103,7 @@
if (time_of_first_rtt_ms_ == -1)
time_of_first_rtt_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
rtt_sum_ms_ += avg_rtt_ms;
+ last_rtt_ms_ = avg_rtt_ms;
++num_rtts_;
}
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 14e2272..cc61080 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -443,6 +443,7 @@
int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
int64_t rtt_sum_ms_ GUARDED_BY(crit_);
+ int64_t last_rtt_ms_ GUARDED_BY(crit_);
size_t num_rtts_ GUARDED_BY(crit_);
// RtpRtcp modules, declared last as they use other members on construction.
