Add histogram stats for send delay for a sent video stream.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
@@ skipping 2038 matching lines @@
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.PaddingBitrateReceivedInKbps"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.RetransmittedBitrateSentInKbps"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
 
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SendSideDelayInMs"));
   EXPECT_EQ(1,
             test::NumHistogramSamples(video_prefix + "SendSideDelayMaxInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SendDelayInMs"));

[stefan-webrtc, 2015/12/10 08:37:50]

Shouldn't we just remove the SendSideDelayInMs above?

[åsapersson, 2015/12/15 14:28:27]

SendSideDelayInMs is currently the delay from capture -> send
I plan to update it to be capture -> sent.

 
   int num_rtx_samples = use_rtx ? 1 : 0;
   EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
       "WebRTC.Video.RtxBitrateSentInKbps"));
   EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
       "WebRTC.Video.RtxBitrateReceivedInKbps"));
 
   int num_red_samples = use_red ? 1 : 0;
   EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
       "WebRTC.Video.FecBitrateSentInKbps"));
@@ skipping 1154 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc