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6ce03592c690134c2ed11b94e900abbc226f058c
platform-external-webrtc/audio/audio_send_stream_unittest.cc
Sebastian Jansson 41f16bec9f Silencing warnings in audio send stream unit tests. 
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.

With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.

Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
2018-02-22 14:26:59 +00:00

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/*
* 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/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_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_METHOD2(OnAllocationLimitsChanged,
void(uint32_t min_send_bitrate_bps,
uint32_t max_padding_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,
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>()),
simulated_clock_(123456),
rtp_transport_(&simulated_clock_, &event_log_, BitrateConstraints()),
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);
if (audio_bwe_enabled) {
EXPECT_CALL(*channel_proxy_,
EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
&rtp_transport_, Ne(nullptr)))
.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_;
SimulatedClock simulated_clock_;
TimeInterval active_lifetime_;
testing::NiceMock<MockRtcEventLog> event_log_;
RtpTransportControllerSend 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,
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
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