NetEq: Fix stats counting in muted mode

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 1590 matching lines @@
   EXPECT_EQ(out_frame_.samples_per_channel_, new_frame.samples_per_channel_);
   EXPECT_EQ(out_frame_.sample_rate_hz_, new_frame.sample_rate_hz_);
   EXPECT_EQ(out_frame_.num_channels_, new_frame.num_channels_);
   EXPECT_EQ(out_frame_.speech_type_, new_frame.speech_type_);
   EXPECT_EQ(out_frame_.vad_activity_, new_frame.vad_activity_);
 
   // Insert new data. Timestamp is corrected for the time elapsed since the last
   // packet. Verify that normal operation resumes.
   InsertPacket(kSamples * counter_);
   GetAudioUntilNormal();
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
+  // concealment samples, in Q14 (16384 = 100%) .The vast majority should be
+  // concealment samples in this test.
+  EXPECT_GT(stats.expand_rate, 14000);
+  // And, it should be greater than the speech_expand_rate.
+  EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
 }
 
 // Verifies that NetEq goes out of muted state when given a delayed packet.
 TEST_F(NetEqDecodingTestWithMutedState, MutedStateDelayedPacket) {
   // Insert one speech packet.
   InsertPacket(0);
   // Pull out audio once and expect it not to be muted.
   EXPECT_FALSE(GetAudioReturnMuted());
   // Pull data until faded out.
   GetAudioUntilMuted();
@@ skipping 148 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc