Add and modify a few ANA stats.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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.
  */
 
@@ skipping 532 matching lines @@
     stats.echo_delay_median_ms = 234;
     stats.echo_delay_std_ms = 567;
     stats.echo_return_loss = 890;
     stats.echo_return_loss_enhancement = 1234;
     stats.residual_echo_likelihood = 0.432f;
     stats.residual_echo_likelihood_recent_max = 0.6f;
     stats.ana_statistics.bitrate_action_counter = rtc::Optional<uint32_t>(321);
     stats.ana_statistics.channel_action_counter = rtc::Optional<uint32_t>(432);
     stats.ana_statistics.dtx_action_counter = rtc::Optional<uint32_t>(543);
     stats.ana_statistics.fec_action_counter = rtc::Optional<uint32_t>(654);
-    stats.ana_statistics.frame_length_action_counter =
+    stats.ana_statistics.frame_length_increase_counter =
         rtc::Optional<uint32_t>(765);
+    stats.ana_statistics.frame_length_decrease_counter =
+        rtc::Optional<uint32_t>(876);
+    stats.ana_statistics.uplink_packet_loss_fraction =
+        rtc::Optional<float>(987.0);
     stats.typing_noise_detected = true;
     return stats;
   }
   void SetAudioSendStreamStats() {
     for (auto* s : call_.GetAudioSendStreams()) {
       s->SetStats(GetAudioSendStreamStats());
     }
   }
   void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info,
                              bool is_sending) {
@@ skipping 19 matching lines @@
     EXPECT_EQ(info.residual_echo_likelihood_recent_max,
               stats.residual_echo_likelihood_recent_max);
     EXPECT_EQ(info.ana_statistics.bitrate_action_counter,
               stats.ana_statistics.bitrate_action_counter);
     EXPECT_EQ(info.ana_statistics.channel_action_counter,
               stats.ana_statistics.channel_action_counter);
     EXPECT_EQ(info.ana_statistics.dtx_action_counter,
               stats.ana_statistics.dtx_action_counter);
     EXPECT_EQ(info.ana_statistics.fec_action_counter,
               stats.ana_statistics.fec_action_counter);
-    EXPECT_EQ(info.ana_statistics.frame_length_action_counter,
-              stats.ana_statistics.frame_length_action_counter);
+    EXPECT_EQ(info.ana_statistics.frame_length_increase_counter,
+              stats.ana_statistics.frame_length_increase_counter);
+    EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter,
+              stats.ana_statistics.frame_length_decrease_counter);
+    EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction,
+              stats.ana_statistics.uplink_packet_loss_fraction);
     EXPECT_EQ(info.typing_noise_detected,
               stats.typing_noise_detected && is_sending);
   }
 
   webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const {
     webrtc::AudioReceiveStream::Stats stats;
     stats.remote_ssrc = 123;
     stats.bytes_rcvd = 456;
     stats.packets_rcvd = 768;
     stats.packets_lost = 101;
@@ skipping 2914 matching lines @@
   // Without this cast, the comparison turned unsigned and, thus, failed for -1.
   const int num_specs = static_cast<int>(specs.size());
   EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
   EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
   EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
 }