Add new decoding statistics for muted output

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 25 matching lines @@
 using testing::ReturnRef;
 
 AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
   AudioDecodingCallStats audio_decode_stats;
   audio_decode_stats.calls_to_silence_generator = 234;
   audio_decode_stats.calls_to_neteq = 567;
   audio_decode_stats.decoded_normal = 890;
   audio_decode_stats.decoded_plc = 123;
   audio_decode_stats.decoded_cng = 456;
   audio_decode_stats.decoded_plc_cng = 789;
+  audio_decode_stats.decoded_muted_output = 987;
   return audio_decode_stats;
 }
 
 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
 const size_t kOneByteExtensionHeaderLength = 4;
 const size_t kOneByteExtensionLength = 4;
 const int kAbsSendTimeId = 2;
 const int kAudioLevelId = 3;
@@ skipping 293 matching lines @@
             stats.accelerate_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
             stats.preemptive_expand_rate);
   EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
             stats.decoding_calls_to_silence_generator);
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
+  EXPECT_EQ(kAudioDecodeStats.decoded_muted_output,
+            stats.decoding_muted_output);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 
 TEST(AudioReceiveStreamTest, SetGain) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state(),
       helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);
 }
 }  // namespace test
 }  // namespace webrtc