commit 6f14be8df8b67feb480f55b3a41e2b8cd06a836d
author asapersson <asapersson@webrtc.org> Mon Nov 16 00:40:49 2015 -0800
committer Commit bot <commit-bot@chromium.org> Mon Nov 16 08:40:57 2015 +0000
tree cc4af6079edeaa6c546a3ad58677552c25544237
parent 3c735f4505f0b5dce12aa1008139d5af7722fd07

Add limit for minimum number of required samples before recording input and sent framerate stats.

BUG=

Review URL: https://codereview.webrtc.org/1446443002

Cr-Commit-Position: refs/heads/master@{#10644}

webrtc/video/end_to_end_tests.cc

diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 6ed3c47..c7133d0 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2003,6 +2003,10 @@
       "WebRTC.Video.DecodedFramesPerSecond"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.RenderFramesPerSecond"));
 
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.OnewayDelayInMs"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
+
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.EncodeTimeInMs"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.DecodeTimeInMs"));
 
@@ -2023,6 +2027,9 @@
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
 
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.SendSideDelayInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.SendSideDelayMaxInMs"));
+
   int num_rtx_samples = use_rtx ? 1 : 0;
   EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
       "WebRTC.Video.RtxBitrateSentInKbps"));

webrtc/video/send_statistics_proxy.cc

diff --git a/webrtc/video/send_statistics_proxy.cc b/webrtc/video/send_statistics_proxy.cc
index 57189f3..5c2052a 100644
--- a/webrtc/video/send_statistics_proxy.cc
+++ b/webrtc/video/send_statistics_proxy.cc
@@ -70,27 +70,22 @@
 }
 
 void SendStatisticsProxy::UpdateHistograms() {
-  int input_fps =
-      round(input_frame_rate_tracker_.ComputeTotalRate());
-  if (input_fps > 0)
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.InputFramesPerSecond", input_fps);
-  int sent_fps =
-      round(sent_frame_rate_tracker_.ComputeTotalRate());
-  if (sent_fps > 0)
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.SentFramesPerSecond", sent_fps);
-
   const int kMinRequiredSamples = 200;
   int in_width = input_width_counter_.Avg(kMinRequiredSamples);
   int in_height = input_height_counter_.Avg(kMinRequiredSamples);
+  int in_fps = round(input_frame_rate_tracker_.ComputeTotalRate());
   if (in_width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputWidthInPixels", in_width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputHeightInPixels", in_height);
+    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.InputFramesPerSecond", in_fps);
   }
   int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
   int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
+  int sent_fps = round(sent_frame_rate_tracker_.ComputeTotalRate());
   if (sent_width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentWidthInPixels", sent_width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentHeightInPixels", sent_height);
+    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.SentFramesPerSecond", sent_fps);
   }
   int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
   if (encode_ms != -1)