Adding stats that can be used to compute output audio levels.

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 39 matching lines @@
 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 kAudioLevelId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kJitterBufferDelay = -7;
 const int kPlayoutBufferDelay = 302;
 const unsigned int kSpeechOutputLevel = 99;
+const double kTotalOutputEnergy = 0.25;
+const double kTotalOutputDuration = 0.5;
+
 const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, 0, -103};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
   ConfigHelper()
@@ skipping 77 matching lines @@
     using testing::DoAll;
     using testing::SetArgPointee;
 
     ASSERT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
         .WillOnce(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
         .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
     EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
         .WillOnce(Return(kSpeechOutputLevel));
+    EXPECT_CALL(*channel_proxy_, GetTotalOutputEnergy())
+        .WillOnce(Return(kTotalOutputEnergy));
+    EXPECT_CALL(*channel_proxy_, GetTotalOutputDuration())
+        .WillOnce(Return(kTotalOutputDuration));
     EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
         .WillOnce(Return(kNetworkStats));
     EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
         .WillOnce(Return(kAudioDecodeStats));
     EXPECT_CALL(*channel_proxy_, GetRecCodec(_))
         .WillOnce(DoAll(SetArgPointee<0>(kCodecInst), Return(true)));
   }
 
  private:
   PacketRouter packet_router_;
@@ skipping 136 matching lines @@
   EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
   EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
   EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000),
             stats.jitter_ms);
   EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
   EXPECT_EQ(kNetworkStats.preferredBufferSize,
             stats.jitter_buffer_preferred_ms);
   EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
             stats.delay_estimate_ms);
   EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+  EXPECT_EQ(kTotalOutputEnergy, stats.total_output_energy);
+  EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
             stats.speech_expand_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
             stats.secondary_decoded_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
             stats.accelerate_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
             stats.preemptive_expand_rate);
   EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
@@ skipping 42 matching lines @@
 
   EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
   EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
   EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
       .WillOnce(Return(true));
 
   recv_stream.Start();
 }
 }  // namespace test
 }  // namespace webrtc