audio/audio_receive_stream_unittest.cc - src - 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 "audio/audio_receive_stream.h"

 #include <map>
 #include <string>
 #include <utility>
 #include <vector>

 #include "api/test/mock_audio_mixer.h"
 #include "api/test/mock_frame_decryptor.h"
 #include "audio/conversion.h"
 #include "audio/mock_voe_channel_proxy.h"
 #include "call/rtp_stream_receiver_controller.h"
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/audio_device/include/mock_audio_device.h"
 #include "modules/audio_processing/include/mock_audio_processing.h"
 #include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "rtc_base/time_utils.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder_factory.h"
 #include "test/mock_transport.h"

 namespace webrtc {
 namespace test {
 namespace {

 using ::testing::_;
 using ::testing::FloatEq;
 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_neteq_plc = 123;
   audio_decode_stats.decoded_codec_plc = 124;
   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 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 int64_t kPlayoutNtpTimestampMs = 5678;

 const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
 const std::pair<int, SdpAudioFormat> kReceiveCodec = {
     123,
     {"codec_name_recv", 96000, 0}};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 789012, 3456, 123, 456, 789, 543, 432,
     321, 123, 101,   0,      {},   789, 12,  345, 678, 901,
     0,   -1,  -1,    -1,     -1,   0,   0,   0,   0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();

 struct ConfigHelper {
   ConfigHelper() : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}

   explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
       : audio_mixer_(audio_mixer) {
     using ::testing::Invoke;

     AudioState::Config config;
     config.audio_mixer = audio_mixer_;
     config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>();
     config.audio_device_module =
         new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>();
     audio_state_ = AudioState::Create(config);

     channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
     EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
     EXPECT_CALL(*channel_receive_,
                 RegisterReceiverCongestionControlObjects(&packet_router_))
         .Times(1);
     EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
         .Times(1);
     EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
     EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
         .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
           EXPECT_THAT(codecs, ::testing::IsEmpty());
         }));

     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     stream_config_.rtcp_send_transport = &rtcp_send_transport_;
     stream_config_.decoder_factory =
         new rtc::RefCountedObject<MockAudioDecoderFactory>;
   }

   std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
     return std::unique_ptr<internal::AudioReceiveStream>(
         new internal::AudioReceiveStream(
             Clock::GetRealTimeClock(), &rtp_stream_receiver_controller_,
             &packet_router_, stream_config_, audio_state_, &event_log_,
             std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_)));
   }

   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
   MockChannelReceive* channel_receive() { return channel_receive_; }

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

     ASSERT_TRUE(channel_receive_);
     EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
         .WillOnce(Return(kCallStats));
     EXPECT_CALL(*channel_receive_, GetDelayEstimate())
         .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
     EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
         .WillOnce(Return(kSpeechOutputLevel));
     EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
         .WillOnce(Return(kTotalOutputEnergy));
     EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
         .WillOnce(Return(kTotalOutputDuration));
     EXPECT_CALL(*channel_receive_, GetNetworkStatistics())
         .WillOnce(Return(kNetworkStats));
     EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
         .WillOnce(Return(kAudioDecodeStats));
     EXPECT_CALL(*channel_receive_, GetReceiveCodec())
         .WillOnce(Return(kReceiveCodec));
     EXPECT_CALL(*channel_receive_, GetCurrentEstimatedPlayoutNtpTimestampMs(_))
         .WillOnce(Return(kPlayoutNtpTimestampMs));
   }

  private:
   PacketRouter packet_router_;
   MockRtcEventLog event_log_;
   rtc::scoped_refptr<AudioState> audio_state_;
   rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
   AudioReceiveStream::Config stream_config_;
   ::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
   RtpStreamReceiverController rtp_stream_receiver_controller_;
   MockTransport rtcp_send_transport_;
 };

 void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
                            int id,
                            uint32_t extension_value,
                            size_t value_length) {
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
   it += 2;

   ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
   it += 2;
   const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
   uint32_t shifted_value = extension_value
                            << (8 * (kExtensionDataLength - value_length));
   *it = (id << 4) + (static_cast<uint8_t>(value_length) - 1);
   ++it;
   ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
                                                              shifted_value);
 }

 const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
     int extension_id,
     uint32_t extension_value,
     size_t value_length) {
   std::vector<uint8_t> header;
   header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
                 kOneByteExtensionLength);
   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.

   BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
                         extension_value, value_length);
   return header;
 }

 const std::vector<uint8_t> CreateRtcpSenderReport() {
   std::vector<uint8_t> packet;
   const size_t kRtcpSrLength = 28;  // In bytes.
   packet.resize(kRtcpSrLength);
   packet[0] = 0x80;  // Version 2.
   packet[1] = 0xc8;  // PT = 200, SR.
   // Length in number of 32-bit words - 1.
   ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
   ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
   return packet;
 }
 }  // namespace

 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
       "{rtp_history_ms: 0}, extensions: [{uri: "
       "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
       "rtcp_send_transport: null, media_transport_config: {media_transport: "
       "null}}",
       config.ToString());
 }

 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();
 }

 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   auto recv_stream = helper.CreateAudioReceiveStream();
   const int kTransportSequenceNumberValue = 1234;
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   constexpr int64_t packet_time_us = 5678000;

   RtpPacketReceived parsed_packet;
   ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
   parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);

   EXPECT_CALL(*helper.channel_receive(),
               OnRtpPacket(::testing::Ref(parsed_packet)));

   recv_stream->OnRtpPacket(parsed_packet);
 }

 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   auto recv_stream = helper.CreateAudioReceiveStream();
   std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
   EXPECT_CALL(*helper.channel_receive(),
               ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
       .WillOnce(Return());
   recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
 }

 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream->GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
   EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
             stats.header_and_padding_bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
   EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
   EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
   EXPECT_EQ(
       kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 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(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
   EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
   EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
   EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
   EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
                 static_cast<double>(rtc::kNumMillisecsPerSec),
             stats.jitter_buffer_delay_seconds);
   EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
             stats.jitter_buffer_emitted_count);
   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.currentSecondaryDiscardedRate),
             stats.secondary_discarded_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_neteq_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_codec_plc, stats.decoding_codec_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);
   EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
 }

 TEST(AudioReceiveStreamTest, SetGain) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();
   EXPECT_CALL(*helper.channel_receive(),
               SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream->SetGain(0.765f);
 }

 TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
   ConfigHelper helper1;
   ConfigHelper helper2(helper1.audio_mixer());
   auto recv_stream1 = helper1.CreateAudioReceiveStream();
   auto recv_stream2 = helper2.CreateAudioReceiveStream();

   EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
   EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
   EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
   EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
   EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
       .WillOnce(Return(true));
   EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
       .WillOnce(Return(true));
   EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get()))
       .Times(1);
   EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get()))
       .Times(1);

   recv_stream1->Start();
   recv_stream2->Start();

   // One more should not result in any more mixer sources added.
   recv_stream1->Start();

   // Stop stream before it is being destructed.
   recv_stream2->Stop();
 }

 TEST(AudioReceiveStreamTest, ReconfigureWithSameConfig) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();
   recv_stream->Reconfigure(helper.config());
 }

 TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();

   auto new_config = helper.config();
   new_config.rtp.nack.rtp_history_ms = 300 + 20;
   new_config.rtp.extensions.clear();
   new_config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
   new_config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                    kTransportSequenceNumberId + 1));
   new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));

   MockChannelReceive& channel_receive = *helper.channel_receive();
   EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
   EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));

   recv_stream->Reconfigure(new_config);
 }

 TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
   ConfigHelper helper;
   auto recv_stream = helper.CreateAudioReceiveStream();

   auto new_config_0 = helper.config();
   rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
       new rtc::RefCountedObject<MockFrameDecryptor>());
   new_config_0.frame_decryptor = mock_frame_decryptor_0;

   recv_stream->Reconfigure(new_config_0);

   auto new_config_1 = helper.config();
   rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
       new rtc::RefCountedObject<MockFrameDecryptor>());
   new_config_1.frame_decryptor = mock_frame_decryptor_1;
   new_config_1.crypto_options.sframe.require_frame_encryption = true;
   recv_stream->Reconfigure(new_config_1);
 }

 }  // 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 "audio/audio_receive_stream.h"

	#include <map>
	#include <string>
	#include <utility>
	#include <vector>

	#include "api/test/mock_audio_mixer.h"
	#include "api/test/mock_frame_decryptor.h"
	#include "audio/conversion.h"
	#include "audio/mock_voe_channel_proxy.h"
	#include "call/rtp_stream_receiver_controller.h"
	#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
	#include "modules/audio_device/include/mock_audio_device.h"
	#include "modules/audio_processing/include/mock_audio_processing.h"
	#include "modules/pacing/packet_router.h"
	#include "modules/rtp_rtcp/source/byte_io.h"
	#include "rtc_base/time_utils.h"
	#include "test/gtest.h"
	#include "test/mock_audio_decoder_factory.h"
	#include "test/mock_transport.h"

	namespace webrtc {
	namespace test {
	namespace {

	using ::testing::_;
	using ::testing::FloatEq;
	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_neteq_plc = 123;
	audio_decode_stats.decoded_codec_plc = 124;
	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 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 int64_t kPlayoutNtpTimestampMs = 5678;

	const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
	const std::pair<int, SdpAudioFormat> kReceiveCodec = {
	123,
	{"codec_name_recv", 96000, 0}};
	const NetworkStatistics kNetworkStats = {
	123, 456, false, 789012, 3456, 123, 456, 789, 543, 432,
	321, 123, 101, 0, {}, 789, 12, 345, 678, 901,
	0, -1, -1, -1, -1, 0, 0, 0, 0};
	const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();

	struct ConfigHelper {
	ConfigHelper() : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}

	explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
	: audio_mixer_(audio_mixer) {
	using ::testing::Invoke;

	AudioState::Config config;
	config.audio_mixer = audio_mixer_;
	config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>();
	config.audio_device_module =
	new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>();
	audio_state_ = AudioState::Create(config);

	channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
	EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
	EXPECT_CALL(*channel_receive_,
	RegisterReceiverCongestionControlObjects(&packet_router_))
	.Times(1);
	EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
	.Times(1);
	EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
	EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
	.WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
	EXPECT_THAT(codecs, ::testing::IsEmpty());
	}));

	stream_config_.rtp.local_ssrc = kLocalSsrc;
	stream_config_.rtp.remote_ssrc = kRemoteSsrc;
	stream_config_.rtp.nack.rtp_history_ms = 300;
	stream_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
	stream_config_.rtp.extensions.push_back(RtpExtension(
	RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
	stream_config_.rtcp_send_transport = &rtcp_send_transport_;
	stream_config_.decoder_factory =
	new rtc::RefCountedObject<MockAudioDecoderFactory>;
	}

	std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
	return std::unique_ptr<internal::AudioReceiveStream>(
	new internal::AudioReceiveStream(
	Clock::GetRealTimeClock(), &rtp_stream_receiver_controller_,
	&packet_router_, stream_config_, audio_state_, &event_log_,
	std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_)));
	}

	AudioReceiveStream::Config& config() { return stream_config_; }
	rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
	MockChannelReceive* channel_receive() { return channel_receive_; }

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

	ASSERT_TRUE(channel_receive_);
	EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
	.WillOnce(Return(kCallStats));
	EXPECT_CALL(*channel_receive_, GetDelayEstimate())
	.WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
	EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
	.WillOnce(Return(kSpeechOutputLevel));
	EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
	.WillOnce(Return(kTotalOutputEnergy));
	EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
	.WillOnce(Return(kTotalOutputDuration));
	EXPECT_CALL(*channel_receive_, GetNetworkStatistics())
	.WillOnce(Return(kNetworkStats));
	EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
	.WillOnce(Return(kAudioDecodeStats));
	EXPECT_CALL(*channel_receive_, GetReceiveCodec())
	.WillOnce(Return(kReceiveCodec));
	EXPECT_CALL(*channel_receive_, GetCurrentEstimatedPlayoutNtpTimestampMs(_))
	.WillOnce(Return(kPlayoutNtpTimestampMs));
	}

	private:
	PacketRouter packet_router_;
	MockRtcEventLog event_log_;
	rtc::scoped_refptr<AudioState> audio_state_;
	rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
	AudioReceiveStream::Config stream_config_;
	::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
	RtpStreamReceiverController rtp_stream_receiver_controller_;
	MockTransport rtcp_send_transport_;
	};

	void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
	int id,
	uint32_t extension_value,
	size_t value_length) {
	const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
	ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
	it += 2;

	ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
	it += 2;
	const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
	uint32_t shifted_value = extension_value
	<< (8 * (kExtensionDataLength - value_length));
	*it = (id << 4) + (static_cast<uint8_t>(value_length) - 1);
	++it;
	ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
	shifted_value);
	}

	const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
	int extension_id,
	uint32_t extension_value,
	size_t value_length) {
	std::vector<uint8_t> header;
	header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
	kOneByteExtensionLength);
	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.

	BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
	extension_value, value_length);
	return header;
	}

	const std::vector<uint8_t> CreateRtcpSenderReport() {
	std::vector<uint8_t> packet;
	const size_t kRtcpSrLength = 28; // In bytes.
	packet.resize(kRtcpSrLength);
	packet[0] = 0x80; // Version 2.
	packet[1] = 0xc8; // PT = 200, SR.
	// Length in number of 32-bit words - 1.
	ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
	ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
	return packet;
	}
	} // namespace

	TEST(AudioReceiveStreamTest, ConfigToString) {
	AudioReceiveStream::Config config;
	config.rtp.remote_ssrc = kRemoteSsrc;
	config.rtp.local_ssrc = kLocalSsrc;
	config.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
	EXPECT_EQ(
	"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
	"{rtp_history_ms: 0}, extensions: [{uri: "
	"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
	"rtcp_send_transport: null, media_transport_config: {media_transport: "
	"null}}",
	config.ToString());
	}

	TEST(AudioReceiveStreamTest, ConstructDestruct) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();
	}

	TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
	ConfigHelper helper;
	helper.config().rtp.transport_cc = true;
	auto recv_stream = helper.CreateAudioReceiveStream();
	const int kTransportSequenceNumberValue = 1234;
	std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
	kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
	constexpr int64_t packet_time_us = 5678000;

	RtpPacketReceived parsed_packet;
	ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
	parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);

	EXPECT_CALL(*helper.channel_receive(),
	OnRtpPacket(::testing::Ref(parsed_packet)));

	recv_stream->OnRtpPacket(parsed_packet);
	}

	TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
	ConfigHelper helper;
	helper.config().rtp.transport_cc = true;
	auto recv_stream = helper.CreateAudioReceiveStream();
	std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
	EXPECT_CALL(*helper.channel_receive(),
	ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
	.WillOnce(Return());
	recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
	}

	TEST(AudioReceiveStreamTest, GetStats) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();
	helper.SetupMockForGetStats();
	AudioReceiveStream::Stats stats = recv_stream->GetStats();
	EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
	EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
	EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
	stats.header_and_padding_bytes_rcvd);
	EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
	stats.packets_rcvd);
	EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
	EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
	EXPECT_EQ(
	kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 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(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
	EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
	EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
	EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
	EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
	static_cast<double>(rtc::kNumMillisecsPerSec),
	stats.jitter_buffer_delay_seconds);
	EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
	stats.jitter_buffer_emitted_count);
	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.currentSecondaryDiscardedRate),
	stats.secondary_discarded_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_neteq_plc, stats.decoding_plc);
	EXPECT_EQ(kAudioDecodeStats.decoded_codec_plc, stats.decoding_codec_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);
	EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
	}

	TEST(AudioReceiveStreamTest, SetGain) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();
	EXPECT_CALL(*helper.channel_receive(),
	SetChannelOutputVolumeScaling(FloatEq(0.765f)));
	recv_stream->SetGain(0.765f);
	}

	TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
	ConfigHelper helper1;
	ConfigHelper helper2(helper1.audio_mixer());
	auto recv_stream1 = helper1.CreateAudioReceiveStream();
	auto recv_stream2 = helper2.CreateAudioReceiveStream();

	EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
	EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
	EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
	EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
	EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
	.WillOnce(Return(true));
	EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
	.WillOnce(Return(true));
	EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get()))
	.Times(1);
	EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get()))
	.Times(1);

	recv_stream1->Start();
	recv_stream2->Start();

	// One more should not result in any more mixer sources added.
	recv_stream1->Start();

	// Stop stream before it is being destructed.
	recv_stream2->Stop();
	}

	TEST(AudioReceiveStreamTest, ReconfigureWithSameConfig) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();
	recv_stream->Reconfigure(helper.config());
	}

	TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();

	auto new_config = helper.config();
	new_config.rtp.nack.rtp_history_ms = 300 + 20;
	new_config.rtp.extensions.clear();
	new_config.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
	new_config.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kTransportSequenceNumberUri,
	kTransportSequenceNumberId + 1));
	new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));

	MockChannelReceive& channel_receive = *helper.channel_receive();
	EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
	EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));

	recv_stream->Reconfigure(new_config);
	}

	TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
	ConfigHelper helper;
	auto recv_stream = helper.CreateAudioReceiveStream();

	auto new_config_0 = helper.config();
	rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
	new rtc::RefCountedObject<MockFrameDecryptor>());
	new_config_0.frame_decryptor = mock_frame_decryptor_0;

	recv_stream->Reconfigure(new_config_0);

	auto new_config_1 = helper.config();
	rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
	new rtc::RefCountedObject<MockFrameDecryptor>());
	new_config_1.frame_decryptor = mock_frame_decryptor_1;
	new_config_1.crypto_options.sframe.require_frame_encryption = true;
	recv_stream->Reconfigure(new_config_1);
	}

	} // namespace test
	} // namespace webrtc