Add encode and decode time to histograms stats:
- "WebRTC.Video.EncodeTimeInMs"
- "WebRTC.Video.DecodeTimeInMs"
BUG=chromium:488243
Review URL: https://codereview.webrtc.org/1250203002
Cr-Commit-Position: refs/heads/master@{#9630}
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 33fc75e..416c8dd 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1661,6 +1661,9 @@
"WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.RenderFramesPerSecond"));
+ EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.EncodeTimeInMs"));
+ EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.DecodeTimeInMs"));
+
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.BitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index e028dab..6604d3d 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -39,13 +39,19 @@
if (render_fps > 0)
RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond", render_fps);
- const int kMinRequiredSamples = 100;
+ const int kMinRequiredSamples = 200;
int width = render_width_counter_.Avg(kMinRequiredSamples);
int height = render_height_counter_.Avg(kMinRequiredSamples);
if (width != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
}
+ // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
+ // not per frame. Change decode time to include every frame.
+ const int kMinRequiredDecodeSamples = 5;
+ int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
+ if (decode_ms != -1)
+ RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
@@ -76,6 +82,7 @@
stats_.jitter_buffer_ms = jitter_buffer_ms;
stats_.min_playout_delay_ms = min_playout_delay_ms;
stats_.render_delay_ms = render_delay_ms;
+ decode_time_counter_.Add(decode_ms);
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
diff --git a/webrtc/video/receive_statistics_proxy.h b/webrtc/video/receive_statistics_proxy.h
index 0d22948..5042210 100644
--- a/webrtc/video/receive_statistics_proxy.h
+++ b/webrtc/video/receive_statistics_proxy.h
@@ -99,6 +99,7 @@
rtc::RateTracker render_fps_tracker_total_ GUARDED_BY(crit_);
SampleCounter render_width_counter_ GUARDED_BY(crit_);
SampleCounter render_height_counter_ GUARDED_BY(crit_);
+ SampleCounter decode_time_counter_ GUARDED_BY(crit_);
ReportBlockStats report_block_stats_ GUARDED_BY(crit_);
};
diff --git a/webrtc/video/send_statistics_proxy.cc b/webrtc/video/send_statistics_proxy.cc
index fe60f1e..551b4b5 100644
--- a/webrtc/video/send_statistics_proxy.cc
+++ b/webrtc/video/send_statistics_proxy.cc
@@ -46,7 +46,7 @@
if (sent_fps > 0)
RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.SentFramesPerSecond", sent_fps);
- const int kMinRequiredSamples = 100;
+ const int kMinRequiredSamples = 200;
int in_width = input_width_counter_.Avg(kMinRequiredSamples);
int in_height = input_height_counter_.Avg(kMinRequiredSamples);
if (in_width != -1) {
@@ -59,6 +59,9 @@
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentWidthInPixels", sent_width);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentHeightInPixels", sent_height);
}
+ int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
+ if (encode_ms != -1)
+ RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.EncodeTimeInMs", encode_ms);
}
void SendStatisticsProxy::OutgoingRate(const int video_channel,
@@ -72,6 +75,7 @@
void SendStatisticsProxy::CpuOveruseMetricsUpdated(
const CpuOveruseMetrics& metrics) {
rtc::CritScope lock(&crit_);
+ // TODO(asapersson): Change to use OnEncodedFrame() for avg_encode_time_ms.
stats_.avg_encode_time_ms = metrics.avg_encode_time_ms;
stats_.encode_usage_percent = metrics.encode_usage_percent;
}
@@ -186,6 +190,11 @@
input_height_counter_.Add(height);
}
+void SendStatisticsProxy::OnEncodedFrame(int encode_time_ms) {
+ rtc::CritScope lock(&crit_);
+ encode_time_counter_.Add(encode_time_ms);
+}
+
void SendStatisticsProxy::RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
diff --git a/webrtc/video/send_statistics_proxy.h b/webrtc/video/send_statistics_proxy.h
index 1d37f18..60d962f 100644
--- a/webrtc/video/send_statistics_proxy.h
+++ b/webrtc/video/send_statistics_proxy.h
@@ -49,6 +49,9 @@
// Used to update incoming frame rate.
void OnIncomingFrame(int width, int height);
+ // Used to update encode time of frames.
+ void OnEncodedFrame(int encode_time_ms);
+
// From VideoEncoderRateObserver.
void OnSetRates(uint32_t bitrate_bps, int framerate) override;
@@ -125,6 +128,7 @@
SampleCounter input_height_counter_ GUARDED_BY(crit_);
SampleCounter sent_width_counter_ GUARDED_BY(crit_);
SampleCounter sent_height_counter_ GUARDED_BY(crit_);
+ SampleCounter encode_time_counter_ GUARDED_BY(crit_);
};
} // namespace webrtc
diff --git a/webrtc/video/video_capture_input.cc b/webrtc/video/video_capture_input.cc
index 1fbadff..456e99c 100644
--- a/webrtc/video/video_capture_input.cc
+++ b/webrtc/video/video_capture_input.cc
@@ -147,8 +147,10 @@
}
// Update the overuse detector with the duration.
if (encode_start_time != -1) {
- overuse_detector_->FrameEncoded(
+ int encode_time_ms = static_cast<int>(
Clock::GetRealTimeClock()->TimeInMilliseconds() - encode_start_time);
+ overuse_detector_->FrameEncoded(encode_time_ms);
+ stats_proxy_->OnEncodedFrame(encode_time_ms);
}
}
// We're done!
