Fetch goog_jitter_received stats for receive stream in WebRTC perf browser test.

chrome/browser/media/webrtc_browsertest_perf.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "chrome/browser/media/webrtc_browsertest_perf.h"
 
 #include "base/strings/stringprintf.h"
 #include "base/values.h"
 #include "chrome/test/base/in_process_browser_test.h"
 #include "testing/perf/perf_test.h"
@@ skipping 17 matching lines @@
     // Not an audio receive stream.
     return false;
   }
 
   EXPECT_TRUE(pc_dict.GetString(Statistic("bytesReceived", ssrc), &value));
   perf_test::PrintResult(
       "audio_bytes", modifier, "bytes_recv", value, "bytes", false);
   EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value));
   perf_test::PrintResult(
       "audio_misc", modifier, "packets_lost", value, "frames", false);
+  EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
+  perf_test::PrintResult(
+      "audio_rx", modifier, "goog_jitter_recv", value, "ms", false);
 
   EXPECT_TRUE(pc_dict.GetString(Statistic("googExpandRate", ssrc), &value));
   perf_test::PrintResult(
       "audio_rates", modifier, "goog_expand_rate", value, "%", false);
   EXPECT_TRUE(
       pc_dict.GetString(Statistic("googSpeechExpandRate", ssrc), &value));
   perf_test::PrintResult(
       "audio_rates", modifier, "goog_speech_expand_rate", value, "%", false);
   EXPECT_TRUE(
       pc_dict.GetString(Statistic("googSecondaryDecodedRate", ssrc), &value));
@@ skipping 11 matching lines @@
   if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) {
     // Not an audio send stream.
     return false;
   }
 
   EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value));
   perf_test::PrintResult(
       "audio_bytes", modifier, "bytes_sent", value, "bytes", false);
   EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
   perf_test::PrintResult(
       "audio_tx", modifier, "goog_jitter_recv", value, "ms", false);

[phoglund_chromium, 2015/11/16 14:06:35]

Then you should remove this, no?

[Solis, 2015/11/16 14:46:15]

Nope, that's the estimated jitter for send streams (as reported back by receiver
through SSRC recever reports).

[phoglund_chromium, 2015/11/16 14:57:58]

We should consider calling this something else though; it's maybe a bit
confusing that there are two goog_jitter_recv in different buckets. Maybe
send_estimated_goog_jitter_recv?...

[Solis, 2015/11/16 15:03:10]

It is confusing for me as well. Which is why I added hta@ to this CL.

Not sure how much of a standard the goog_ stats are, how swiftly they can be
changed and such...

   EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), &value));
   perf_test::PrintResult(
       "audio_tx", modifier, "goog_rtt", value, "ms", false);
   return true;
 }
 
 static bool MaybePrintResultsForVideoSend(
     const std::string& ssrc, const base::DictionaryValue& pc_dict,
     const std::string& modifier) {
   std::string value;
@@ skipping 166 matching lines @@
         MaybePrintResultsForAudioSend(ssrc, pc_dict, modifier) ||
         MaybePrintResultsForVideoReceive(ssrc, pc_dict, modifier) ||
         MaybePrintResultsForVideoSend(ssrc, pc_dict, modifier);
     ASSERT_TRUE(did_recognize_stream_type) << "Failed to figure out which "
                                               "kind of stream SSRC " << ssrc
                                            << " is. ";
   }
 }
 
 }  // namespace test