audio/audio_receive_stream_unittest.cc - src/webrtc - Git at Google

 /*
  *  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.
  */

 #include <string>

 #include "testing/gtest/include/gtest/gtest.h"

 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"

 namespace webrtc {
 namespace test {
 namespace {

 using testing::_;
 using testing::Return;

 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;
   return audio_decode_stats;
 }

 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
 const size_t kAbsoluteSendTimeLength = 4;
 const int kAbsSendTimeId = 2;
 const int kAudioLevelId = 3;
 const int kJitterBufferDelay = -7;
 const int kPlayoutBufferDelay = 302;
 const unsigned int kSpeechOutputLevel = 99;
 const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, -198, -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() {
     using testing::Invoke;

     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);

     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
           EXPECT_FALSE(channel_proxy_);
           channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
                   .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
   }

   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }

   void SetupMockForGetStats() {
     using testing::DoAll;
     using testing::SetArgReferee;

     EXPECT_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_, GetNetworkStatistics())
         .WillOnce(Return(kNetworkStats));
     EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
         .WillOnce(Return(kAudioDecodeStats));

     EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
         .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
   }

  private:
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };

 void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
                                     int id,
                                     uint32_t abs_send_time) {
   const size_t kRtpOneByteHeaderLength = 4;
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);

   const uint32_t kPosLength = 2;
   ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
                                        kAbsoluteSendTimeLength / 4);

   const uint8_t kLengthOfData = 3;
   buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
   ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
       buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
 }

 size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
                                       int extension_id,
                                       uint32_t abs_send_time) {
   header[0] = 0x80;   // Version 2.
   header[0] |= 0x10;  // Set extension bit.
   header[1] = 100;    // Payload type.
   header[1] |= 0x80;  // Marker bit is set.
   ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234);  // Sequence number.
   ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678);  // Timestamp.
   ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321);  // SSRC.
   int32_t rtp_header_length = webrtc::kRtpHeaderSize;

   BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
                                  abs_send_time);
   rtp_header_length += kAbsoluteSendTimeLength;
   return rtp_header_length;
 }
 }  // namespace

 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.voe_channel_id = kChannelId;
   config.combined_audio_video_bwe = true;
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
       "receive_transport: nullptr, rtcp_send_transport: nullptr, "
       "voe_channel_id: 2, combined_audio_video_bwe: true}",
       config.ToString());
 }

 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
 }

 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
   ConfigHelper helper;
   helper.config().combined_audio_video_bwe = true;
   internal::AudioReceiveStream recv_stream(
       helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
   uint8_t rtp_packet[30];
   const int kAbsSendTimeValue = 1234;
   CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              sizeof(rtp_packet) - kExpectedHeaderLength,
                              testing::_, false))
       .Times(1);
   EXPECT_TRUE(
       recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
 }

 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
   EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
   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(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,
             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(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc
	/*
	* 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.
	*/

	#include <string>

	#include "testing/gtest/include/gtest/gtest.h"

	#include "webrtc/audio/audio_receive_stream.h"
	#include "webrtc/audio/conversion.h"
	#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
	#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
	#include "webrtc/test/mock_voe_channel_proxy.h"
	#include "webrtc/test/mock_voice_engine.h"

	namespace webrtc {
	namespace test {
	namespace {

	using testing::_;
	using testing::Return;

	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;
	return audio_decode_stats;
	}

	const int kChannelId = 2;
	const uint32_t kRemoteSsrc = 1234;
	const uint32_t kLocalSsrc = 5678;
	const size_t kAbsoluteSendTimeLength = 4;
	const int kAbsSendTimeId = 2;
	const int kAudioLevelId = 3;
	const int kJitterBufferDelay = -7;
	const int kPlayoutBufferDelay = 302;
	const unsigned int kSpeechOutputLevel = 99;
	const CallStatistics kCallStats = {
	345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123};
	const CodecInst kCodecInst = {
	123, "codec_name_recv", 96000, -187, -198, -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() {
	using testing::Invoke;

	EXPECT_CALL(voice_engine_,
	RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
	EXPECT_CALL(voice_engine_,
	DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
	AudioState::Config config;
	config.voice_engine = &voice_engine_;
	audio_state_ = AudioState::Create(config);

	EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
	.WillOnce(Invoke([this](int channel_id) {
	EXPECT_FALSE(channel_proxy_);
	channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
	EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
	EXPECT_CALL(*channel_proxy_,
	SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
	.Times(1);
	EXPECT_CALL(*channel_proxy_,
	SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
	.Times(1);
	return channel_proxy_;
	}));
	stream_config_.voe_channel_id = kChannelId;
	stream_config_.rtp.local_ssrc = kLocalSsrc;
	stream_config_.rtp.remote_ssrc = kRemoteSsrc;
	stream_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
	stream_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
	}

	MockRemoteBitrateEstimator* remote_bitrate_estimator() {
	return &remote_bitrate_estimator_;
	}
	AudioReceiveStream::Config& config() { return stream_config_; }
	rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
	MockVoiceEngine& voice_engine() { return voice_engine_; }

	void SetupMockForGetStats() {
	using testing::DoAll;
	using testing::SetArgReferee;

	EXPECT_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_, GetNetworkStatistics())
	.WillOnce(Return(kNetworkStats));
	EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
	.WillOnce(Return(kAudioDecodeStats));

	EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
	.WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
	}

	private:
	MockRemoteBitrateEstimator remote_bitrate_estimator_;
	testing::StrictMock<MockVoiceEngine> voice_engine_;
	rtc::scoped_refptr<AudioState> audio_state_;
	AudioReceiveStream::Config stream_config_;
	testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
	};

	void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
	int id,
	uint32_t abs_send_time) {
	const size_t kRtpOneByteHeaderLength = 4;
	const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
	ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);

	const uint32_t kPosLength = 2;
	ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
	kAbsoluteSendTimeLength / 4);

	const uint8_t kLengthOfData = 3;
	buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
	ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
	buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
	}

	size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
	int extension_id,
	uint32_t abs_send_time) {
	header[0] = 0x80; // Version 2.
	header[0] \|= 0x10; // Set extension bit.
	header[1] = 100; // Payload type.
	header[1] \|= 0x80; // Marker bit is set.
	ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number.
	ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp.
	ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC.
	int32_t rtp_header_length = webrtc::kRtpHeaderSize;

	BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
	abs_send_time);
	rtp_header_length += kAbsoluteSendTimeLength;
	return rtp_header_length;
	}
	} // namespace

	TEST(AudioReceiveStreamTest, ConfigToString) {
	AudioReceiveStream::Config config;
	config.rtp.remote_ssrc = kRemoteSsrc;
	config.rtp.local_ssrc = kLocalSsrc;
	config.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
	config.voe_channel_id = kChannelId;
	config.combined_audio_video_bwe = true;
	EXPECT_EQ(
	"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
	"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
	"receive_transport: nullptr, rtcp_send_transport: nullptr, "
	"voe_channel_id: 2, combined_audio_video_bwe: true}",
	config.ToString());
	}

	TEST(AudioReceiveStreamTest, ConstructDestruct) {
	ConfigHelper helper;
	internal::AudioReceiveStream recv_stream(
	helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
	}

	TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
	ConfigHelper helper;
	helper.config().combined_audio_video_bwe = true;
	internal::AudioReceiveStream recv_stream(
	helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
	uint8_t rtp_packet[30];
	const int kAbsSendTimeValue = 1234;
	CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
	PacketTime packet_time(5678000, 0);
	const size_t kExpectedHeaderLength = 20;
	EXPECT_CALL(*helper.remote_bitrate_estimator(),
	IncomingPacket(packet_time.timestamp / 1000,
	sizeof(rtp_packet) - kExpectedHeaderLength,
	testing::_, false))
	.Times(1);
	EXPECT_TRUE(
	recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
	}

	TEST(AudioReceiveStreamTest, GetStats) {
	ConfigHelper helper;
	internal::AudioReceiveStream recv_stream(
	helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
	helper.SetupMockForGetStats();
	AudioReceiveStream::Stats stats = recv_stream.GetStats();
	EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
	EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
	EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
	stats.packets_rcvd);
	EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
	EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
	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(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,
	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(kCallStats.capture_start_ntp_time_ms_,
	stats.capture_start_ntp_time_ms);
	}
	} // namespace test
	} // namespace webrtc