Only update codec type histogram if lifetime is long enough (10 sec).
Add metrics for Call/VideoSendStream/VideoReceiveStream lifetime.
BUG=
Review-Url: https://codereview.webrtc.org/2136533002
Cr-Commit-Position: refs/heads/master@{#13537}
diff --git a/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java b/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java
index 14ca30b..510c895 100644
--- a/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java
+++ b/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java
@@ -744,13 +744,13 @@
answeringExpectations.expectStateChange(DataChannel.State.CLOSED);
answeringExpectations.dataChannel.close();
offeringExpectations.dataChannel.close();
- getMetrics();
// Free the Java-land objects and collect them.
shutdownPC(offeringPC, offeringExpectations);
offeringPC = null;
shutdownPC(answeringPC, answeringExpectations);
answeringPC = null;
+ getMetrics();
videoSource.dispose();
factory.dispose();
System.gc();
@@ -1026,13 +1026,11 @@
private static void getMetrics() {
Metrics metrics = Metrics.getAndReset();
assertTrue(metrics.map.size() > 0);
- // Test for example that the configured video codec is recorded when a
- // VideoSendStream is created.
- String name = "WebRTC.Video.Encoder.CodecType";
+ // Test for example that the lifetime of a Call is recorded.
+ String name = "WebRTC.Call.LifetimeInSeconds";
assertTrue(metrics.map.containsKey(name));
HistogramInfo info = metrics.map.get(name);
- assertEquals(1, info.samples.size()); // samples: <sample value, # of events>
- assertTrue(info.samples.containsValue(2)); // <codec type, 2>, same codec configured
+ assertTrue(info.samples.size() > 0);
}
private static void shutdownPC(
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 6df86e3..f50a8d6 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -136,6 +136,7 @@
void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
void UpdateReceiveHistograms();
+ void UpdateHistograms();
void UpdateAggregateNetworkState();
Clock* const clock_;
@@ -196,6 +197,7 @@
VieRemb remb_;
const std::unique_ptr<CongestionController> congestion_controller_;
const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
+ const int64_t start_ms_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
@@ -231,11 +233,11 @@
min_allocated_send_bitrate_bps_(0),
num_bitrate_updates_(0),
configured_max_padding_bitrate_bps_(0),
-
remb_(clock_),
congestion_controller_(
new CongestionController(clock_, this, &remb_, event_log_.get())),
- video_send_delay_stats_(new SendDelayStats(clock_)) {
+ video_send_delay_stats_(new SendDelayStats(clock_)),
+ start_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
@@ -285,10 +287,17 @@
// they won't try to concurrently update stats.
UpdateSendHistograms();
UpdateReceiveHistograms();
+ UpdateHistograms();
Trace::ReturnTrace();
}
+void Call::UpdateHistograms() {
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+ "WebRTC.Call.LifetimeInSeconds",
+ (clock_->TimeInMilliseconds() - start_ms_) / 1000);
+}
+
void Call::UpdateSendHistograms() {
if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
return;
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index d0e99ab..7e25f95 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2134,12 +2134,17 @@
screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
// Verify that stats have been updated once.
+ EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
+ EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
+ EXPECT_EQ(1,
+ metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
+
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index 57d1f56..9057543 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -24,6 +24,7 @@
Clock* clock)
: clock_(clock),
config_(*config),
+ start_ms_(clock->TimeInMilliseconds()),
// 1000ms window, scale 1000 for ms to s.
decode_fps_estimator_(1000, 1000),
renders_fps_estimator_(1000, 1000),
@@ -39,6 +40,10 @@
}
void ReceiveStatisticsProxy::UpdateHistograms() {
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+ "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
+ (clock_->TimeInMilliseconds() - start_ms_) / 1000);
+
int fraction_lost = report_block_stats_.FractionLostInPercent();
if (fraction_lost != -1) {
RTC_LOGGED_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
diff --git a/webrtc/video/receive_statistics_proxy.h b/webrtc/video/receive_statistics_proxy.h
index 86c35e7..09c5f12 100644
--- a/webrtc/video/receive_statistics_proxy.h
+++ b/webrtc/video/receive_statistics_proxy.h
@@ -104,6 +104,7 @@
// ReceiveStatisticsProxy() ctor to accept a const& of Config (since we'll
// then no longer store a pointer to the object).
const VideoReceiveStream::Config& config_;
+ const int64_t start_ms_;
rtc::CriticalSection crit_;
VideoReceiveStream::Stats stats_ GUARDED_BY(crit_);
diff --git a/webrtc/video/send_statistics_proxy.cc b/webrtc/video/send_statistics_proxy.cc
index 8852de1..dfa7185f 100644
--- a/webrtc/video/send_statistics_proxy.cc
+++ b/webrtc/video/send_statistics_proxy.cc
@@ -77,16 +77,23 @@
: clock_(clock),
config_(config),
content_type_(content_type),
+ start_ms_(clock->TimeInMilliseconds()),
last_sent_frame_timestamp_(0),
encode_time_(kEncodeTimeWeigthFactor),
uma_container_(
new UmaSamplesContainer(GetUmaPrefix(content_type_), stats_, clock)) {
- UpdateCodecTypeHistogram(config_.encoder_settings.payload_name);
}
SendStatisticsProxy::~SendStatisticsProxy() {
rtc::CritScope lock(&crit_);
uma_container_->UpdateHistograms(config_, stats_);
+
+ int64_t elapsed_sec = (clock_->TimeInMilliseconds() - start_ms_) / 1000;
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Video.SendStreamLifetimeInSeconds",
+ elapsed_sec);
+
+ if (elapsed_sec >= metrics::kMinRunTimeInSeconds)
+ UpdateCodecTypeHistogram(config_.encoder_settings.payload_name);
}
SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
diff --git a/webrtc/video/send_statistics_proxy.h b/webrtc/video/send_statistics_proxy.h
index 2c9225f..b47691a 100644
--- a/webrtc/video/send_statistics_proxy.h
+++ b/webrtc/video/send_statistics_proxy.h
@@ -139,6 +139,7 @@
const VideoSendStream::Config config_;
rtc::CriticalSection crit_;
VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_);
+ const int64_t start_ms_;
VideoSendStream::Stats stats_ GUARDED_BY(crit_);
uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
diff --git a/webrtc/video/send_statistics_proxy_unittest.cc b/webrtc/video/send_statistics_proxy_unittest.cc
index ad51f2a..b1bf161 100644
--- a/webrtc/video/send_statistics_proxy_unittest.cc
+++ b/webrtc/video/send_statistics_proxy_unittest.cc
@@ -302,6 +302,21 @@
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InputWidthInPixels"));
}
+TEST_F(SendStatisticsProxyTest, LifetimeHistogramIsUpdated) {
+ const int64_t kTimeSec = 3;
+ fake_clock_.AdvanceTimeMilliseconds(kTimeSec * 1000);
+ statistics_proxy_.reset();
+ EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
+ EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.SendStreamLifetimeInSeconds",
+ kTimeSec));
+}
+
+TEST_F(SendStatisticsProxyTest, CodecTypeHistogramIsUpdated) {
+ fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
+ statistics_proxy_.reset();
+ EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.Encoder.CodecType"));
+}
+
TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_Vp8) {
EncodedImage encoded_image;
CodecSpecificInfo codec_info;
