blob: d8ff0fd083144efe7d5da60c3b50c7ca7a1bbef4 [file] [log] [blame]
/*
* 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 <utility>
#include <vector>
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
#include "call/test/mock_rtp_transport_controller_send.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "rtc_base/fakeclock.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/timedelta.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace test {
namespace {
using testing::_;
using testing::Eq;
using testing::Ne;
using testing::Invoke;
using testing::Return;
using testing::StrEq;
const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
const int kTransportSequenceNumberId = 4;
const int32_t kEchoDelayMedian = 254;
const int32_t kEchoDelayStdDev = -3;
const double kDivergentFilterFraction = 0.2f;
const double kEchoReturnLoss = -65;
const double kEchoReturnLossEnhancement = 101;
const double kResidualEchoLikelihood = -1.0f;
const double kResidualEchoLikelihoodMax = 23.0f;
const CallStatistics kCallStats = {
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const int kTelephoneEventPayloadFrequency = 65432;
const int kTelephoneEventCode = 45;
const int kTelephoneEventDuration = 6789;
const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000};
constexpr int kIsacPayloadType = 103;
const SdpAudioFormat kIsacFormat = {"isac", 16000, 1};
const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
const SdpAudioFormat kG722Format = {"g722", 8000, 1};
const AudioCodecSpec kCodecSpecs[] = {
{kIsacFormat, {16000, 1, 32000, 10000, 32000}},
{kOpusFormat, {48000, 1, 32000, 6000, 510000}},
{kG722Format, {16000, 1, 64000}}};
class MockLimitObserver : public BitrateAllocator::LimitObserver {
public:
MOCK_METHOD3(OnAllocationLimitsChanged,
void(uint32_t min_send_bitrate_bps,
uint32_t max_padding_bitrate_bps,
uint32_t total_bitrate_bps));
};
std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
int payload_type,
const SdpAudioFormat& format) {
for (const auto& spec : kCodecSpecs) {
if (format == spec.format) {
std::unique_ptr<MockAudioEncoder> encoder(
new testing::NiceMock<MockAudioEncoder>());
ON_CALL(*encoder.get(), SampleRateHz())
.WillByDefault(Return(spec.info.sample_rate_hz));
ON_CALL(*encoder.get(), NumChannels())
.WillByDefault(Return(spec.info.num_channels));
ON_CALL(*encoder.get(), RtpTimestampRateHz())
.WillByDefault(Return(spec.format.clockrate_hz));
return encoder;
}
}
return nullptr;
}
rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
rtc::scoped_refptr<MockAudioEncoderFactory> factory =
new rtc::RefCountedObject<MockAudioEncoderFactory>();
ON_CALL(*factory.get(), GetSupportedEncoders())
.WillByDefault(Return(std::vector<AudioCodecSpec>(
std::begin(kCodecSpecs), std::end(kCodecSpecs))));
ON_CALL(*factory.get(), QueryAudioEncoder(_))
.WillByDefault(Invoke(
[](const SdpAudioFormat& format) -> rtc::Optional<AudioCodecInfo> {
for (const auto& spec : kCodecSpecs) {
if (format == spec.format) {
return spec.info;
}
}
return rtc::nullopt;
}));
ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
.WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
rtc::Optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioEncoder>* return_value) {
*return_value = SetupAudioEncoderMock(payload_type, format);
}));
return factory;
}
struct ConfigHelper {
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: stream_config_(nullptr),
audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
bitrate_allocator_(&limit_observer_),
worker_queue_("ConfigHelper_worker_queue"),
audio_encoder_(nullptr) {
using testing::Invoke;
AudioState::Config config;
config.audio_mixer = AudioMixerImpl::Create();
config.audio_processing = audio_processing_;
config.audio_device_module =
new rtc::RefCountedObject<MockAudioDeviceModule>();
audio_state_ = AudioState::Create(config);
SetupDefaultChannelProxy(audio_bwe_enabled);
SetupMockForSetupSendCodec(expect_set_encoder_call);
// Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
// calls from the default ctor behavior.
stream_config_.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
stream_config_.rtp.ssrc = kSsrc;
stream_config_.rtp.nack.rtp_history_ms = 200;
stream_config_.rtp.c_name = kCName;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
if (audio_bwe_enabled) {
AddBweToConfig(&stream_config_);
}
stream_config_.encoder_factory = SetupEncoderFactoryMock();
stream_config_.min_bitrate_bps = 10000;
stream_config_.max_bitrate_bps = 65000;
}
std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
return std::unique_ptr<internal::AudioSendStream>(
new internal::AudioSendStream(
stream_config_, audio_state_, &worker_queue_, &rtp_transport_,
&bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, rtc::nullopt,
&active_lifetime_,
std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
}
AudioSendStream::Config& config() { return stream_config_; }
MockAudioEncoderFactory& mock_encoder_factory() {
return *static_cast<MockAudioEncoderFactory*>(
stream_config_.encoder_factory.get());
}
MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
TimeInterval* active_lifetime() { return &active_lifetime_; }
static void AddBweToConfig(AudioSendStream::Config* config) {
config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberId));
config->send_codec_spec->transport_cc_enabled = true;
}
void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
EXPECT_TRUE(channel_proxy_ == nullptr);
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
.WillRepeatedly(Invoke(
[this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
*rtp_rtcp_module = &this->rtp_rtcp_;
*rtp_receiver = nullptr; // Not deemed necessary for tests yet.
}));
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
.WillRepeatedly(Return(&bandwidth_observer_));
if (audio_bwe_enabled) {
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
RegisterSenderCongestionControlObjects(
&rtp_transport_, Eq(&bandwidth_observer_)))
.Times(1);
} else {
EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
&rtp_transport_, Eq(nullptr)))
.Times(1);
}
EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
.Times(1);
{
::testing::InSequence unregister_on_destruction;
EXPECT_CALL(*channel_proxy_, RegisterTransport(_)).Times(1);
EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(1);
}
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1); // Destructor resets the event log
EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull()))
.Times(1); // Destructor resets the rtt stats.
}
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
if (expect_set_encoder_call) {
EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _))
.WillOnce(Invoke(
[this](int payload_type, std::unique_ptr<AudioEncoder>* encoder) {
this->audio_encoder_ = std::move(*encoder);
return true;
}));
}
}
void SetupMockForModifyEncoder() {
// Let ModifyEncoder to invoke mock audio encoder.
EXPECT_CALL(*channel_proxy_, ModifyEncoder(_))
.WillRepeatedly(Invoke(
[this](rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) {
if (this->audio_encoder_)
modifier(&this->audio_encoder_);
}));
}
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_,
SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency))
.WillOnce(Return(true));
EXPECT_CALL(*channel_proxy_,
SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
.WillOnce(Return(true));
}
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgPointee;
using testing::SetArgReferee;
std::vector<ReportBlock> report_blocks;
webrtc::ReportBlock block = kReportBlock;
report_blocks.push_back(block); // Has wrong SSRC.
block.source_SSRC = kSsrc;
report_blocks.push_back(block); // Correct block.
block.fraction_lost = 0;
report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost.
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
.WillRepeatedly(Return(kCallStats));
EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_proxy_, GetANAStatistics())
.WillRepeatedly(Return(ANAStats()));
audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
audio_processing_stats_.echo_return_loss_enhancement =
kEchoReturnLossEnhancement;
audio_processing_stats_.delay_median_ms = kEchoDelayMedian;
audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev;
audio_processing_stats_.divergent_filter_fraction =
kDivergentFilterFraction;
audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood;
audio_processing_stats_.residual_echo_likelihood_recent_max =
kResidualEchoLikelihoodMax;
EXPECT_CALL(*audio_processing_, GetStatistics(true))
.WillRepeatedly(Return(audio_processing_stats_));
}
private:
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
AudioProcessingStats audio_processing_stats_;
TimeInterval active_lifetime_;
testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
testing::NiceMock<MockRtcEventLog> event_log_;
testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
testing::NiceMock<MockRtpRtcp> rtp_rtcp_;
MockRtcpRttStats rtcp_rtt_stats_;
testing::NiceMock<MockLimitObserver> limit_observer_;
BitrateAllocator bitrate_allocator_;
// |worker_queue| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
rtc::TaskQueue worker_queue_;
std::unique_ptr<AudioEncoder> audio_encoder_;
};
} // namespace
TEST(AudioSendStreamTest, ConfigToString) {
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.c_name = kCName;
config.min_bitrate_bps = 12000;
config.max_bitrate_bps = 34000;
config.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
config.send_codec_spec->nack_enabled = true;
config.send_codec_spec->transport_cc_enabled = false;
config.send_codec_spec->cng_payload_type = 42;
config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
EXPECT_EQ(
"{rtp: {ssrc: 1234, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
"{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, "
"min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"cng_payload_type: 42, payload_type: 103, "
"format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
"parameters: {}}}}",
config.ToString());
}
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream->SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency, kTelephoneEventCode,
kTelephoneEventDuration));
}
TEST(AudioSendStreamTest, SetMuted) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
send_stream->SetMuted(true);
}
TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(true, true);
auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream->GetStats(true);
EXPECT_EQ(kSsrc, stats.local_ssrc);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
EXPECT_EQ(std::string(kIsacCodec.plname), stats.codec_name);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
stats.ext_seqnum);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
(kIsacCodec.plfreq / 1000)),
stats.jitter_ms);
EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
EXPECT_EQ(0, stats.audio_level);
EXPECT_EQ(0, stats.total_input_energy);
EXPECT_EQ(0, stats.total_input_duration);
EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms);
EXPECT_EQ(kEchoDelayStdDev, stats.apm_statistics.delay_standard_deviation_ms);
EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss);
EXPECT_EQ(kEchoReturnLossEnhancement,
stats.apm_statistics.echo_return_loss_enhancement);
EXPECT_EQ(kDivergentFilterFraction,
stats.apm_statistics.divergent_filter_fraction);
EXPECT_EQ(kResidualEchoLikelihood,
stats.apm_statistics.residual_echo_likelihood);
EXPECT_EQ(kResidualEchoLikelihoodMax,
stats.apm_statistics.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.typing_noise_detected);
}
TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
ConfigHelper helper(false, true);
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
const std::string kAnaConfigString = "abcde";
const std::string kAnaReconfigString = "12345";
helper.config().audio_network_adaptor_config = kAnaConfigString;
EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
.WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
int payload_type, const SdpAudioFormat& format,
rtc::Optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioEncoder>* return_value) {
auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
EXPECT_CALL(*mock_encoder,
EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
.WillOnce(Return(true));
EXPECT_CALL(*mock_encoder,
EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _))
.WillOnce(Return(true));
*return_value = std::move(mock_encoder);
}));
auto send_stream = helper.CreateAudioSendStream();
auto stream_config = helper.config();
stream_config.audio_network_adaptor_config = kAnaReconfigString;
helper.SetupMockForModifyEncoder();
send_stream->Reconfigure(stream_config);
}
// VAD is applied when codec is mono and the CNG frequency matches the codec
// clock rate.
TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
ConfigHelper helper(false, false);
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
helper.config().send_codec_spec->cng_payload_type = 105;
using ::testing::Invoke;
std::unique_ptr<AudioEncoder> stolen_encoder;
EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
.WillOnce(
Invoke([&stolen_encoder](int payload_type,
std::unique_ptr<AudioEncoder>* encoder) {
stolen_encoder = std::move(*encoder);
return true;
}));
auto send_stream = helper.CreateAudioSendStream();
// We cannot truly determine if the encoder created is an AudioEncoderCng. It
// is the only reasonable implementation that will return something from
// ReclaimContainedEncoders, though.
ASSERT_TRUE(stolen_encoder);
EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty());
}
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(),
SetBitrate(helper.config().max_bitrate_bps, _));
send_stream->OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
6000);
}
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
send_stream->OnBitrateUpdated(50000, 0.0, 50, 5000);
}
// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
TEST(AudioSendStreamTest, DontRecreateEncoder) {
ConfigHelper helper(false, false);
// WillOnce is (currently) the default used by ConfigHelper if asked to set an
// expectation for SetEncoder. Since this behavior is essential for this test
// to be correct, it's instead set-up manually here. Otherwise a simple change
// to ConfigHelper (say to WillRepeatedly) would silently make this test
// useless.
EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
.WillOnce(Return(true));
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
helper.config().send_codec_spec->cng_payload_type = 105;
auto send_stream = helper.CreateAudioSendStream();
send_stream->Reconfigure(helper.config());
}
TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
ConfigHelper::AddBweToConfig(&new_config);
EXPECT_CALL(*helper.channel_proxy(),
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
{
::testing::InSequence seq;
EXPECT_CALL(*helper.channel_proxy(), ResetSenderCongestionControlObjects())
.Times(1);
EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects(
helper.transport(), Ne(nullptr)))
.Times(1);
}
send_stream->Reconfigure(new_config);
}
// Checks that AudioSendStream logs the times at which RTP packets are sent
// through its interface.
TEST(AudioSendStreamTest, UpdateLifetime) {
ConfigHelper helper(false, true);
MockTransport mock_transport;
helper.config().send_transport = &mock_transport;
Transport* registered_transport;
ON_CALL(*helper.channel_proxy(), RegisterTransport(_))
.WillByDefault(Invoke([&registered_transport](Transport* transport) {
registered_transport = transport;
}));
rtc::ScopedFakeClock fake_clock;
constexpr int64_t kTimeBetweenSendRtpCallsMs = 100;
{
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(mock_transport, SendRtp(_, _, _)).Times(2);
const PacketOptions options;
registered_transport->SendRtp(nullptr, 0, options);
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kTimeBetweenSendRtpCallsMs));
registered_transport->SendRtp(nullptr, 0, options);
}
EXPECT_TRUE(!helper.active_lifetime()->Empty());
EXPECT_EQ(helper.active_lifetime()->Length(), kTimeBetweenSendRtpCallsMs);
}
} // namespace test
} // namespace webrtc