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Move some integration test functions to the .cc file

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The integration_test_helpers.h file was too long and had too many
big functions inline.

This CL takes some of the largest and puts them in the .cc file.

Bug: None
Change-Id: Ibaaf9675ca8b5efa29878b4883b21f14104451a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349020
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42169}
This commit is contained in:
Harald Alvestrand 2024-04-24 12:09:39 +00:00 committed by WebRTC LUCI CQ
parent f9a5ed0399
commit 1a3120f3fd
2 changed files with 169 additions and 154 deletions
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166
pc/test/integration_test_helpers.cc
View File

@ -93,4 +93,170 @@ TimeDelta TaskQueueMetronome::TickPeriod() const {
return tick_period_;
}
// Implementation of PeerConnectionIntegrationWrapper functions
void PeerConnectionIntegrationWrapper::StartWatchingDelayStats() {
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = NewGetStats();
auto rtp_stats =
received_stats->GetStatsOfType<RTCInboundRtpStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.has_value());
ASSERT_TRUE(rtp_stats->packets_received.has_value());
rtp_stats_id_ = rtp_stats->id();
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
void PeerConnectionIntegrationWrapper::UpdateDelayStats(std::string tag,
int desc_size) {
auto report = NewGetStats();
auto rtp_stats = report->GetAs<RTCInboundRtpStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =
*rtp_stats->relative_packet_arrival_delay - audio_delay_stat_;
auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
// The purpose of these checks is to sound the alarm early if we introduce
// serious regressions. The numbers are not acceptable for production, but
// occur on slow bots.
//
// An average relative packet arrival delay over the renegotiation of
// > 100 ms indicates that something is dramatically wrong, and will impact
// quality for sure.
// Worst bots:
// linux_x86_dbg at 0.206
#if !defined(NDEBUG)
EXPECT_GT(0.25, recent_delay) << tag << " size " << desc_size;
#else
EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size;
#endif
auto delta_samples = *rtp_stats->total_samples_received - audio_samples_stat_;
auto delta_concealed = *rtp_stats->concealed_samples - audio_concealed_stat_;
// These limits should be adjusted down as we improve:
//
// Concealing more than 4000 samples during a renegotiation is unacceptable.
// But some bots are slow.
// Worst bots:
// linux_more_configs bot at conceal count 5184
// android_arm_rel at conceal count 9241
// linux_x86_dbg at 15174
#if !defined(NDEBUG)
EXPECT_GT(18000U, delta_concealed) << "Concealed " << delta_concealed
<< " of " << delta_samples << " samples";
#else
EXPECT_GT(15000U, delta_concealed) << "Concealed " << delta_concealed
<< " of " << delta_samples << " samples";
#endif
// Concealing more than 20% of samples during a renegotiation is
// unacceptable.
// Worst bots:
// Nondebug: Linux32 Release at conceal rate 0.606597 (CI run)
// Debug: linux_x86_dbg bot at conceal rate 0.854
// internal bot at conceal rate 0.967 (b/294020344)
// TODO(https://crbug.com/webrtc/15393): Improve audio quality during
// renegotiation so that we can reduce these thresholds, 99% is not even
// close to the 20% deemed unacceptable above or the 0% that would be ideal.
if (delta_samples > 0) {
#if !defined(NDEBUG)
EXPECT_LT(1.0 * delta_concealed / delta_samples, 0.99)
<< "Concealed " << delta_concealed << " of " << delta_samples
<< " samples";
#else
EXPECT_LT(1.0 * delta_concealed / delta_samples, 0.7)
<< "Concealed " << delta_concealed << " of " << delta_samples
<< " samples";
#endif
}
// Increment trailing counters
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
bool PeerConnectionIntegrationWrapper::Init(
const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
PeerConnectionDependencies dependencies,
rtc::SocketServer* socket_server,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
bool create_media_engine) {
// There's an error in this test code if Init ends up being called twice.
RTC_DCHECK(!peer_connection_);
RTC_DCHECK(!peer_connection_factory_);
fake_network_manager_.reset(new rtc::FakeNetworkManager());
fake_network_manager_->AddInterface(kDefaultLocalAddress);
socket_factory_.reset(new rtc::BasicPacketSocketFactory(socket_server));
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::BasicPortAllocator(fake_network_manager_.get(),
socket_factory_.get()));
port_allocator_ = port_allocator.get();
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (!fake_audio_capture_module_) {
return false;
}
rtc::Thread* const signaling_thread = rtc::Thread::Current();
PeerConnectionFactoryDependencies pc_factory_dependencies;
pc_factory_dependencies.network_thread = network_thread;
pc_factory_dependencies.worker_thread = worker_thread;
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
pc_factory_dependencies.decode_metronome =
std::make_unique<TaskQueueMetronome>(TimeDelta::Millis(8));
pc_factory_dependencies.adm = fake_audio_capture_module_;
if (create_media_engine) {
EnableMediaWithDefaults(pc_factory_dependencies);
}
if (reset_encoder_factory) {
pc_factory_dependencies.video_encoder_factory.reset();
}
if (reset_decoder_factory) {
pc_factory_dependencies.video_decoder_factory.reset();
}
if (!pc_factory_dependencies.audio_processing) {
// If the standard Creation method for APM returns a null pointer, instead
// use the builder for testing to create an APM object.
pc_factory_dependencies.audio_processing =
AudioProcessingBuilderForTesting().Create();
}
if (event_log_factory) {
event_log_factory_ = event_log_factory.get();
pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
} else {
pc_factory_dependencies.event_log_factory =
std::make_unique<RtcEventLogFactory>();
}
peer_connection_factory_ =
CreateModularPeerConnectionFactory(std::move(pc_factory_dependencies));
if (!peer_connection_factory_) {
return false;
}
if (options) {
peer_connection_factory_->SetOptions(*options);
}
if (config) {
sdp_semantics_ = config->sdp_semantics;
}
dependencies.allocator = std::move(port_allocator);
peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
return peer_connection_.get() != nullptr;
}
} // namespace webrtc

157
pc/test/integration_test_helpers.h
View File

@ -645,88 +645,9 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
}
// Functions for querying stats.
void StartWatchingDelayStats() {
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = NewGetStats();
auto rtp_stats =
received_stats->GetStatsOfType<RTCInboundRtpStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.has_value());
ASSERT_TRUE(rtp_stats->packets_received.has_value());
rtp_stats_id_ = rtp_stats->id();
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
void StartWatchingDelayStats();
void UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
auto rtp_stats = report->GetAs<RTCInboundRtpStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =
*rtp_stats->relative_packet_arrival_delay - audio_delay_stat_;
auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
// The purpose of these checks is to sound the alarm early if we introduce
// serious regressions. The numbers are not acceptable for production, but
// occur on slow bots.
//
// An average relative packet arrival delay over the renegotiation of
// > 100 ms indicates that something is dramatically wrong, and will impact
// quality for sure.
// Worst bots:
// linux_x86_dbg at 0.206
#if !defined(NDEBUG)
EXPECT_GT(0.25, recent_delay) << tag << " size " << desc_size;
#else
EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size;
#endif
auto delta_samples =
*rtp_stats->total_samples_received - audio_samples_stat_;
auto delta_concealed =
*rtp_stats->concealed_samples - audio_concealed_stat_;
// These limits should be adjusted down as we improve:
//
// Concealing more than 4000 samples during a renegotiation is unacceptable.
// But some bots are slow.
// Worst bots:
// linux_more_configs bot at conceal count 5184
// android_arm_rel at conceal count 9241
// linux_x86_dbg at 15174
#if !defined(NDEBUG)
EXPECT_GT(18000U, delta_concealed) << "Concealed " << delta_concealed
<< " of " << delta_samples << " samples";
#else
EXPECT_GT(15000U, delta_concealed) << "Concealed " << delta_concealed
<< " of " << delta_samples << " samples";
#endif
// Concealing more than 20% of samples during a renegotiation is
// unacceptable.
// Worst bots:
// Nondebug: Linux32 Release at conceal rate 0.606597 (CI run)
// Debug: linux_x86_dbg bot at conceal rate 0.854
// internal bot at conceal rate 0.967 (b/294020344)
// TODO(https://crbug.com/webrtc/15393): Improve audio quality during
// renegotiation so that we can reduce these thresholds, 99% is not even
// close to the 20% deemed unacceptable above or the 0% that would be ideal.
if (delta_samples > 0) {
#if !defined(NDEBUG)
EXPECT_LT(1.0 * delta_concealed / delta_samples, 0.99)
<< "Concealed " << delta_concealed << " of " << delta_samples
<< " samples";
#else
EXPECT_LT(1.0 * delta_concealed / delta_samples, 0.7)
<< "Concealed " << delta_concealed << " of " << delta_samples
<< " samples";
#endif
}
// Increment trailing counters
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
void UpdateDelayStats(std::string tag, int desc_size);
// Sets number of candidates expected
void ExpectCandidates(int candidate_count) {
@ -747,79 +668,7 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
bool create_media_engine) {
// There's an error in this test code if Init ends up being called twice.
RTC_DCHECK(!peer_connection_);
RTC_DCHECK(!peer_connection_factory_);
fake_network_manager_.reset(new rtc::FakeNetworkManager());
fake_network_manager_->AddInterface(kDefaultLocalAddress);
socket_factory_.reset(new rtc::BasicPacketSocketFactory(socket_server));
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::BasicPortAllocator(fake_network_manager_.get(),
socket_factory_.get()));
port_allocator_ = port_allocator.get();
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (!fake_audio_capture_module_) {
return false;
}
rtc::Thread* const signaling_thread = rtc::Thread::Current();
PeerConnectionFactoryDependencies pc_factory_dependencies;
pc_factory_dependencies.network_thread = network_thread;
pc_factory_dependencies.worker_thread = worker_thread;
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.task_queue_factory =
CreateDefaultTaskQueueFactory();
pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
pc_factory_dependencies.decode_metronome =
std::make_unique<TaskQueueMetronome>(TimeDelta::Millis(8));
pc_factory_dependencies.adm = fake_audio_capture_module_;
if (create_media_engine) {
EnableMediaWithDefaults(pc_factory_dependencies);
}
if (reset_encoder_factory) {
pc_factory_dependencies.video_encoder_factory.reset();
}
if (reset_decoder_factory) {
pc_factory_dependencies.video_decoder_factory.reset();
}
if (!pc_factory_dependencies.audio_processing) {
// If the standard Creation method for APM returns a null pointer, instead
// use the builder for testing to create an APM object.
pc_factory_dependencies.audio_processing =
AudioProcessingBuilderForTesting().Create();
}
if (event_log_factory) {
event_log_factory_ = event_log_factory.get();
pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
} else {
pc_factory_dependencies.event_log_factory =
std::make_unique<RtcEventLogFactory>();
}
peer_connection_factory_ =
CreateModularPeerConnectionFactory(std::move(pc_factory_dependencies));
if (!peer_connection_factory_) {
return false;
}
if (options) {
peer_connection_factory_->SetOptions(*options);
}
if (config) {
sdp_semantics_ = config->sdp_semantics;
}
dependencies.allocator = std::move(port_allocator);
peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
return peer_connection_.get() != nullptr;
}
bool create_media_engine);
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration* config,