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webrtc/pc/test/integration_test_helpers.h
Henrik Boström 2fb83072db Move more non-standard metrics to inbound-rtp. 
They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.

To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.

Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
2022-10-07 07:22:04 +00:00

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/*
* Copyright 2012 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.
*/
#ifndef PC_TEST_INTEGRATION_TEST_HELPERS_H_
#define PC_TEST_INTEGRATION_TEST_HELPERS_H_
#include <limits.h>
#include <stdint.h>
#include <stdio.h>
#include <algorithm>
#include <functional>
#include <limits>
#include <list>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/call/call_factory_interface.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/data_channel_interface.h"
#include "api/field_trials_view.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/uma_metrics.h"
#include "api/units/time_delta.h"
#include "api/video/video_rotation.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "media/engine/fake_webrtc_video_engine.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "p2p/base/fake_ice_transport.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/mock_async_resolver.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/port_interface.h"
#include "p2p/base/test_stun_server.h"
#include "p2p/base/test_turn_customizer.h"
#include "p2p/base/test_turn_server.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/dtmf_sender.h"
#include "pc/local_audio_source.h"
#include "pc/media_session.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory.h"
#include "pc/peer_connection_proxy.h"
#include "pc/rtp_media_utils.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_periodic_video_source.h"
#include "pc/test/fake_periodic_video_track_source.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/fake_video_track_renderer.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "pc/video_track_source.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/fake_mdns_responder.h"
#include "rtc_base/fake_network.h"
#include "rtc_base/firewall_socket_server.h"
#include "rtc_base/gunit.h"
#include "rtc_base/helpers.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/logging.h"
#include "rtc_base/mdns_responder_interface.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/test_certificate_verifier.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
#include "test/scoped_key_value_config.h"
namespace webrtc {
using ::cricket::ContentInfo;
using ::cricket::StreamParams;
using ::rtc::SocketAddress;
using ::testing::_;
using ::testing::Combine;
using ::testing::Contains;
using ::testing::DoAll;
using ::testing::ElementsAre;
using ::testing::NiceMock;
using ::testing::Return;
using ::testing::SetArgPointee;
using ::testing::UnorderedElementsAreArray;
using ::testing::Values;
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
static const int kDefaultTimeout = 10000;
static const int kMaxWaitForStatsMs = 3000;
static const int kMaxWaitForActivationMs = 5000;
static const int kMaxWaitForFramesMs = 10000;
// Default number of audio/video frames to wait for before considering a test
// successful.
static const int kDefaultExpectedAudioFrameCount = 3;
static const int kDefaultExpectedVideoFrameCount = 3;
static const char kDataChannelLabel[] = "data_channel";
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
static const int kDefaultSrtpCryptoSuite = rtc::kSrtpAes128CmSha1_80;
static const int kDefaultSrtpCryptoSuiteGcm = rtc::kSrtpAeadAes256Gcm;
static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
// Helper function for constructing offer/answer options to initiate an ICE
// restart.
PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions();
// Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
// attribute from received SDP, simulating a legacy endpoint.
void RemoveSsrcsAndMsids(cricket::SessionDescription* desc);
// Removes all stream information besides the stream ids, simulating an
// endpoint that only signals a=msid lines to convey stream_ids.
void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
// TODO(https://crbug.com/webrtc/14175): Stop depending on "track" stats, the
// metrics we're interested in are already available in "inbound-rtp".
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
media_stats_vec);
class TaskQueueMetronome : public webrtc::Metronome {
public:
TaskQueueMetronome(TaskQueueFactory* factory, TimeDelta tick_period);
~TaskQueueMetronome() override;
// webrtc::Metronome implementation.
void AddListener(TickListener* listener) override;
void RemoveListener(TickListener* listener) override;
TimeDelta TickPeriod() const override;
private:
Mutex mutex_;
const TimeDelta tick_period_;
std::set<TickListener*> listeners_ RTC_GUARDED_BY(mutex_);
RepeatingTaskHandle tick_task_;
rtc::TaskQueue queue_;
};
class SignalingMessageReceiver {
public:
virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0;
virtual void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) = 0;
protected:
SignalingMessageReceiver() {}
virtual ~SignalingMessageReceiver() {}
};
class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
public:
explicit MockRtpReceiverObserver(cricket::MediaType media_type)
: expected_media_type_(media_type) {}
void OnFirstPacketReceived(cricket::MediaType media_type) override {
ASSERT_EQ(expected_media_type_, media_type);
first_packet_received_ = true;
}
bool first_packet_received() const { return first_packet_received_; }
virtual ~MockRtpReceiverObserver() {}
private:
bool first_packet_received_ = false;
cricket::MediaType expected_media_type_;
};
// Helper class that wraps a peer connection, observes it, and can accept
// signaling messages from another wrapper.
//
// Uses a fake network, fake A/V capture, and optionally fake
// encoders/decoders, though they aren't used by default since they don't
// advertise support of any codecs.
// TODO(steveanton): See how this could become a subclass of
// PeerConnectionWrapper defined in peerconnectionwrapper.h.
class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
public SignalingMessageReceiver {
public:
webrtc::PeerConnectionFactoryInterface* pc_factory() const {
return peer_connection_factory_.get();
}
webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
// If a signaling message receiver is set (via ConnectFakeSignaling), this
// will set the whole offer/answer exchange in motion. Just need to wait for
// the signaling state to reach "stable".
void CreateAndSetAndSignalOffer() {
auto offer = CreateOfferAndWait();
ASSERT_NE(nullptr, offer);
EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
}
// Sets the options to be used when CreateAndSetAndSignalOffer is called, or
// when a remote offer is received (via fake signaling) and an answer is
// generated. By default, uses default options.
void SetOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
offer_answer_options_ = options;
}
// Set a callback to be invoked when SDP is received via the fake signaling
// channel, which provides an opportunity to munge (modify) the SDP. This is
// used to test SDP being applied that a PeerConnection would normally not
// generate, but a non-JSEP endpoint might.
void SetReceivedSdpMunger(
std::function<void(cricket::SessionDescription*)> munger) {
received_sdp_munger_ = std::move(munger);
}
// Similar to the above, but this is run on SDP immediately after it's
// generated.
void SetGeneratedSdpMunger(
std::function<void(cricket::SessionDescription*)> munger) {
generated_sdp_munger_ = std::move(munger);
}
// Set a callback to be invoked when a remote offer is received via the fake
// signaling channel. This provides an opportunity to change the
// PeerConnection state before an answer is created and sent to the caller.
void SetRemoteOfferHandler(std::function<void()> handler) {
remote_offer_handler_ = std::move(handler);
}
void SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) {
remote_async_resolver_ = resolver;
}
// Every ICE connection state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::IceConnectionState>
ice_connection_state_history() const {
return ice_connection_state_history_;
}
void clear_ice_connection_state_history() {
ice_connection_state_history_.clear();
}
// Every standardized ICE connection state in order that has been seen by the
// observer.
std::vector<PeerConnectionInterface::IceConnectionState>
standardized_ice_connection_state_history() const {
return standardized_ice_connection_state_history_;
}
// Every PeerConnection state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::PeerConnectionState>
peer_connection_state_history() const {
return peer_connection_state_history_;
}
// Every ICE gathering state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::IceGatheringState>
ice_gathering_state_history() const {
return ice_gathering_state_history_;
}
std::vector<cricket::CandidatePairChangeEvent>
ice_candidate_pair_change_history() const {
return ice_candidate_pair_change_history_;
}
// Every PeerConnection signaling state in order that has been seen by the
// observer.
std::vector<PeerConnectionInterface::SignalingState>
peer_connection_signaling_state_history() const {
return peer_connection_signaling_state_history_;
}
void AddAudioVideoTracks() {
AddAudioTrack();
AddVideoTrack();
}
rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
return AddTrack(CreateLocalAudioTrack());
}
rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
return AddTrack(CreateLocalVideoTrack());
}
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
cricket::AudioOptions options;
// Disable highpass filter so that we can get all the test audio frames.
options.highpass_filter = false;
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
peer_connection_factory_->CreateAudioSource(options);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
source.get());
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
webrtc::FakePeriodicVideoSource::Config config;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithConfig(
webrtc::FakePeriodicVideoSource::Config config) {
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
webrtc::FakePeriodicVideoSource::Config config;
config.rotation = rotation;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids = {}) {
EXPECT_TRUE(track);
if (!track) {
return nullptr;
}
auto result = pc()->AddTrack(track, stream_ids);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
if (result.ok()) {
return result.MoveValue();
} else {
return nullptr;
}
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
cricket::MediaType media_type) {
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
for (const auto& receiver : pc()->GetReceivers()) {
if (receiver->media_type() == media_type) {
receivers.push_back(receiver);
}
}
return receivers;
}
rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
cricket::MediaType media_type) {
for (auto transceiver : pc()->GetTransceivers()) {
if (transceiver->receiver()->media_type() == media_type) {
return transceiver;
}
}
return nullptr;
}
bool SignalingStateStable() {
return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
}
bool IceGatheringStateComplete() {
return pc()->ice_gathering_state() ==
webrtc::PeerConnectionInterface::kIceGatheringComplete;
}
void CreateDataChannel() { CreateDataChannel(nullptr); }
void CreateDataChannel(const webrtc::DataChannelInit* init) {
CreateDataChannel(kDataChannelLabel, init);
}
void CreateDataChannel(const std::string& label,
const webrtc::DataChannelInit* init) {
auto data_channel_or_error = pc()->CreateDataChannelOrError(label, init);
ASSERT_TRUE(data_channel_or_error.ok());
data_channels_.push_back(data_channel_or_error.MoveValue());
ASSERT_TRUE(data_channels_.back().get() != nullptr);
data_observers_.push_back(
std::make_unique<MockDataChannelObserver>(data_channels_.back().get()));
}
// Return the last observed data channel.
DataChannelInterface* data_channel() {
if (data_channels_.size() == 0) {
return nullptr;
}
return data_channels_.back().get();
}
// Return all data channels.
std::vector<rtc::scoped_refptr<DataChannelInterface>>& data_channels() {
return data_channels_;
}
const MockDataChannelObserver* data_observer() const {
if (data_observers_.size() == 0) {
return nullptr;
}
return data_observers_.back().get();
}
std::vector<std::unique_ptr<MockDataChannelObserver>>& data_observers() {
return data_observers_;
}
int audio_frames_received() const {
return fake_audio_capture_module_->frames_received();
}
// Takes minimum of video frames received for each track.
//
// Can be used like:
// EXPECT_GE(expected_frames, min_video_frames_received_per_track());
//
// To ensure that all video tracks received at least a certain number of
// frames.
int min_video_frames_received_per_track() const {
int min_frames = INT_MAX;
if (fake_video_renderers_.empty()) {
return 0;
}
for (const auto& pair : fake_video_renderers_) {
min_frames = std::min(min_frames, pair.second->num_rendered_frames());
}
return min_frames;
}
// Returns a MockStatsObserver in a state after stats gathering finished,
// which can be used to access the gathered stats.
rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
webrtc::MediaStreamTrackInterface* track) {
auto observer = rtc::make_ref_counted<MockStatsObserver>();
EXPECT_TRUE(peer_connection_->GetStats(
observer.get(), nullptr,
PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer;
}
// Version that doesn't take a track "filter", and gathers all stats.
rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
return OldGetStatsForTrack(nullptr);
}
// Synchronously gets stats and returns them. If it times out, fails the test
// and returns null.
rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
auto callback =
rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
peer_connection_->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
}
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? 0
: fake_video_renderers_.begin()->second->width();
}
int rendered_height() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? 0
: fake_video_renderers_.begin()->second->height();
}
double rendered_aspect_ratio() {
if (rendered_height() == 0) {
return 0.0;
}
return static_cast<double>(rendered_width()) / rendered_height();
}
webrtc::VideoRotation rendered_rotation() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? webrtc::kVideoRotation_0
: fake_video_renderers_.begin()->second->rotation();
}
int local_rendered_width() {
return local_video_renderer_ ? local_video_renderer_->width() : 0;
}
int local_rendered_height() {
return local_video_renderer_ ? local_video_renderer_->height() : 0;
}
double local_rendered_aspect_ratio() {
if (local_rendered_height() == 0) {
return 0.0;
}
return static_cast<double>(local_rendered_width()) /
local_rendered_height();
}
size_t number_of_remote_streams() {
if (!pc()) {
return 0;
}
return pc()->remote_streams()->count();
}
StreamCollectionInterface* remote_streams() const {
if (!pc()) {
ADD_FAILURE();
return nullptr;
}
return pc()->remote_streams().get();
}
StreamCollectionInterface* local_streams() {
if (!pc()) {
ADD_FAILURE();
return nullptr;
}
return pc()->local_streams().get();
}
webrtc::PeerConnectionInterface::SignalingState signaling_state() {
return pc()->signaling_state();
}
webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
return pc()->ice_connection_state();
}
webrtc::PeerConnectionInterface::IceConnectionState
standardized_ice_connection_state() {
return pc()->standardized_ice_connection_state();
}
webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
return pc()->ice_gathering_state();
}
// Returns a MockRtpReceiverObserver for each RtpReceiver returned by
// GetReceivers. They're updated automatically when a remote offer/answer
// from the fake signaling channel is applied, or when
// ResetRtpReceiverObservers below is called.
const std::vector<std::unique_ptr<MockRtpReceiverObserver>>&
rtp_receiver_observers() {
return rtp_receiver_observers_;
}
void ResetRtpReceiverObservers() {
rtp_receiver_observers_.clear();
for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver :
pc()->GetReceivers()) {
std::unique_ptr<MockRtpReceiverObserver> observer(
new MockRtpReceiverObserver(receiver->media_type()));
receiver->SetObserver(observer.get());
rtp_receiver_observers_.push_back(std::move(observer));
}
}
rtc::FakeNetworkManager* network_manager() const {
return fake_network_manager_.get();
}
cricket::PortAllocator* port_allocator() const { return port_allocator_; }
webrtc::FakeRtcEventLogFactory* event_log_factory() const {
return event_log_factory_;
}
const cricket::Candidate& last_candidate_gathered() const {
return last_candidate_gathered_;
}
const cricket::IceCandidateErrorEvent& error_event() const {
return error_event_;
}
// Sets the mDNS responder for the owned fake network manager and keeps a
// reference to the responder.
void SetMdnsResponder(
std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) {
RTC_DCHECK(mdns_responder != nullptr);
mdns_responder_ = mdns_responder.get();
network_manager()->set_mdns_responder(std::move(mdns_responder));
}
// Returns null on failure.
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndWait() {
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
pc()->CreateOffer(observer.get(), offer_answer_options_);
return WaitForDescriptionFromObserver(observer.get());
}
bool Rollback() {
return SetRemoteDescription(
webrtc::CreateSessionDescription(SdpType::kRollback, ""));
}
// 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<webrtc::RTCInboundRTPStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
ASSERT_TRUE(rtp_stats->packets_received.is_defined());
ASSERT_TRUE(rtp_stats->track_id.is_defined());
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 UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
auto rtp_stats =
report->GetAs<webrtc::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
if (delta_samples > 0) {
#if !defined(NDEBUG)
EXPECT_LT(1.0 * delta_concealed / delta_samples, 0.95)
<< "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;
}
// Sets number of candidates expected
void ExpectCandidates(int candidate_count) {
candidates_expected_ = candidate_count;
}
private:
// Constructor used by friend class PeerConnectionIntegrationBaseTest.
explicit PeerConnectionIntegrationWrapper(const std::string& debug_name)
: debug_name_(debug_name) {}
bool Init(const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
rtc::SocketServer* socket_server,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
std::unique_ptr<webrtc::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);
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::BasicPortAllocator(
fake_network_manager_.get(),
std::make_unique<rtc::BasicPacketSocketFactory>(socket_server)));
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();
webrtc::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 =
webrtc::CreateDefaultTaskQueueFactory();
pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
pc_factory_dependencies.metronome = std::make_unique<TaskQueueMetronome>(
pc_factory_dependencies.task_queue_factory.get(), TimeDelta::Millis(8));
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory =
pc_factory_dependencies.task_queue_factory.get();
media_deps.adm = fake_audio_capture_module_;
webrtc::SetMediaEngineDefaults(&media_deps);
if (reset_encoder_factory) {
media_deps.video_encoder_factory.reset();
}
if (reset_decoder_factory) {
media_deps.video_decoder_factory.reset();
}
if (!media_deps.audio_processing) {
// If the standard Creation method for APM returns a null pointer, instead
// use the builder for testing to create an APM object.
media_deps.audio_processing = AudioProcessingBuilderForTesting().Create();
}
media_deps.trials = pc_factory_dependencies.trials.get();
if (create_media_engine) {
pc_factory_dependencies.media_engine =
cricket::CreateMediaEngine(std::move(media_deps));
}
pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
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<webrtc::RtcEventLogFactory>(
pc_factory_dependencies.task_queue_factory.get());
}
peer_connection_factory_ = webrtc::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;
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
PeerConnectionInterface::RTCConfiguration modified_config;
modified_config.sdp_semantics = sdp_semantics_;
// If `config` is null, this will result in a default configuration being
// used.
if (config) {
modified_config = *config;
}
// Disable resolution adaptation; we don't want it interfering with the
// test results.
// TODO(deadbeef): Do something more robust. Since we're testing for aspect
// ratios and not specific resolutions, is this even necessary?
modified_config.set_cpu_adaptation(false);
dependencies.observer = this;
auto peer_connection_or_error =
peer_connection_factory_->CreatePeerConnectionOrError(
modified_config, std::move(dependencies));
return peer_connection_or_error.ok() ? peer_connection_or_error.MoveValue()
: nullptr;
}
void set_signaling_message_receiver(
SignalingMessageReceiver* signaling_message_receiver) {
signaling_message_receiver_ = signaling_message_receiver;
}
void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
void set_signal_ice_candidates(bool signal) {
signal_ice_candidates_ = signal;
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
webrtc::FakePeriodicVideoSource::Config config) {
// Set max frame rate to 10fps to reduce the risk of test flakiness.
// TODO(deadbeef): Do something more robust.
config.frame_interval_ms = 100;
video_track_sources_.emplace_back(
rtc::make_ref_counted<webrtc::FakePeriodicVideoTrackSource>(
config, false /* remote */));
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
peer_connection_factory_->CreateVideoTrack(
rtc::CreateRandomUuid(), video_track_sources_.back().get()));
if (!local_video_renderer_) {
local_video_renderer_.reset(
new webrtc::FakeVideoTrackRenderer(track.get()));
}
return track;
}
void HandleIncomingOffer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
std::unique_ptr<SessionDescriptionInterface> desc =
webrtc::CreateSessionDescription(SdpType::kOffer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
// Setting a remote description may have changed the number of receivers,
// so reset the receiver observers.
ResetRtpReceiverObservers();
if (remote_offer_handler_) {
remote_offer_handler_();
}
auto answer = CreateAnswer();
ASSERT_NE(nullptr, answer);
EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
}
void HandleIncomingAnswer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
std::unique_ptr<SessionDescriptionInterface> desc =
webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
// Set the RtpReceiverObserver after receivers are created.
ResetRtpReceiverObservers();
}
// Returns null on failure.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
pc()->CreateAnswer(observer.get(), offer_answer_options_);
return WaitForDescriptionFromObserver(observer.get());
}
std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
MockCreateSessionDescriptionObserver* observer) {
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (!observer->result()) {
return nullptr;
}
auto description = observer->MoveDescription();
if (generated_sdp_munger_) {
generated_sdp_munger_(description->description());
}
return description;
}
// Setting the local description and sending the SDP message over the fake
// signaling channel are combined into the same method because the SDP
// message needs to be sent as soon as SetLocalDescription finishes, without
// waiting for the observer to be called. This ensures that ICE candidates
// don't outrace the description.
bool SetLocalDescriptionAndSendSdpMessage(
std::unique_ptr<SessionDescriptionInterface> desc) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
SdpType type = desc->GetType();
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp;
pc()->SetLocalDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
// As mentioned above, we need to send the message immediately after
// SetLocalDescription.
SendSdpMessage(type, sdp);
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return true;
}
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
pc()->SetRemoteDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer->result();
}
// This is a work around to remove unused fake_video_renderers from
// transceivers that have either stopped or are no longer receiving.
void RemoveUnusedVideoRenderers() {
if (sdp_semantics_ != SdpSemantics::kUnifiedPlan) {
return;
}
auto transceivers = pc()->GetTransceivers();
std::set<std::string> active_renderers;
for (auto& transceiver : transceivers) {
// Note - we don't check for direction here. This function is called
// before direction is set, and in that case, we should not remove
// the renderer.
if (transceiver->receiver()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
active_renderers.insert(transceiver->receiver()->track()->id());
}
}
for (auto it = fake_video_renderers_.begin();
it != fake_video_renderers_.end();) {
// Remove fake video renderers belonging to any non-active transceivers.
if (!active_renderers.count(it->first)) {
it = fake_video_renderers_.erase(it);
} else {
it++;
}
}
}
// Simulate sending a blob of SDP with delay `signaling_delay_ms_` (0 by
// default).
void SendSdpMessage(SdpType type, const std::string& msg) {
if (signaling_delay_ms_ == 0) {
RelaySdpMessageIfReceiverExists(type, msg);
} else {
rtc::Thread::Current()->PostDelayedTask(
SafeTask(task_safety_.flag(),
[this, type, msg] {
RelaySdpMessageIfReceiverExists(type, msg);
}),
TimeDelta::Millis(signaling_delay_ms_));
}
}
void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveSdpMessage(type, msg);
}
}
// Simulate trickling an ICE candidate with delay `signaling_delay_ms_` (0 by
// default).
void SendIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
if (signaling_delay_ms_ == 0) {
RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg);
} else {
rtc::Thread::Current()->PostDelayedTask(
SafeTask(task_safety_.flag(),
[this, sdp_mid, sdp_mline_index, msg] {
RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index,
msg);
}),
TimeDelta::Millis(signaling_delay_ms_));
}
}
void RelayIceMessageIfReceiverExists(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
msg);
}
}
// SignalingMessageReceiver callbacks.
void ReceiveSdpMessage(SdpType type, const std::string& msg) override {
if (type == SdpType::kOffer) {
HandleIncomingOffer(msg);
} else {
HandleIncomingAnswer(msg);
}
}
void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) override {
RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
absl::optional<RTCError> result;
pc()->AddIceCandidate(absl::WrapUnique(webrtc::CreateIceCandidate(
sdp_mid, sdp_mline_index, msg, nullptr)),
[&result](RTCError r) { result = r; });
EXPECT_TRUE_WAIT(result.has_value(), kDefaultTimeout);
EXPECT_TRUE(result.value().ok());
}
// PeerConnectionObserver callbacks.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
peer_connection_signaling_state_history_.push_back(new_state);
}
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
rtc::scoped_refptr<VideoTrackInterface> video_track(
static_cast<VideoTrackInterface*>(receiver->track().get()));
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
fake_video_renderers_.end());
fake_video_renderers_[video_track->id()] =
std::make_unique<FakeVideoTrackRenderer>(video_track.get());
}
}
void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
auto it = fake_video_renderers_.find(receiver->track()->id());
if (it != fake_video_renderers_.end()) {
fake_video_renderers_.erase(it);
} else {
RTC_LOG(LS_ERROR) << "OnRemoveTrack called for non-active renderer";
}
}
}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
EXPECT_EQ(pc()->ice_connection_state(), new_state);
ice_connection_state_history_.push_back(new_state);
}
void OnStandardizedIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
standardized_ice_connection_state_history_.push_back(new_state);
}
void OnConnectionChange(
webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
peer_connection_state_history_.push_back(new_state);
}
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
EXPECT_EQ(pc()->ice_gathering_state(), new_state);
ice_gathering_state_history_.push_back(new_state);
}
void OnIceSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event) {
ice_candidate_pair_change_history_.push_back(event);
}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
if (remote_async_resolver_) {
const auto& local_candidate = candidate->candidate();
if (local_candidate.address().IsUnresolvedIP()) {
RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE);
rtc::SocketAddress resolved_addr(local_candidate.address());
const auto resolved_ip = mdns_responder_->GetMappedAddressForName(
local_candidate.address().hostname());
RTC_DCHECK(!resolved_ip.IsNil());
resolved_addr.SetResolvedIP(resolved_ip);
EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _))
.WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true)));
EXPECT_CALL(*remote_async_resolver_, Destroy(_));
}
}
// Check if we expected to have a candidate.
EXPECT_GT(candidates_expected_, 1);
candidates_expected_--;
std::string ice_sdp;
EXPECT_TRUE(candidate->ToString(&ice_sdp));
if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) {
// Remote party may be deleted.
return;
}
SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
last_candidate_gathered_ = candidate->candidate();
}
void OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text) override {
error_event_ = cricket::IceCandidateErrorEvent(address, port, url,
error_code, error_text);
}
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
data_channels_.push_back(data_channel);
data_observers_.push_back(
std::make_unique<MockDataChannelObserver>(data_channel.get()));
}
std::string debug_name_;
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
// Reference to the mDNS responder owned by `fake_network_manager_` after set.
webrtc::FakeMdnsResponder* mdns_responder_ = nullptr;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
cricket::PortAllocator* port_allocator_;
// Needed to keep track of number of frames sent.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
fake_video_renderers_;
// Needed to ensure frames aren't received for removed tracks.
std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
removed_fake_video_renderers_;
// For remote peer communication.
SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
int signaling_delay_ms_ = 0;
bool signal_ice_candidates_ = true;
cricket::Candidate last_candidate_gathered_;
cricket::IceCandidateErrorEvent error_event_;
// Store references to the video sources we've created, so that we can stop
// them, if required.
std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
video_track_sources_;
// `local_video_renderer_` attached to the first created local video track.
std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
SdpSemantics sdp_semantics_;
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
std::function<void()> remote_offer_handler_;
rtc::MockAsyncResolver* remote_async_resolver_ = nullptr;
// All data channels either created or observed on this peerconnection
std::vector<rtc::scoped_refptr<DataChannelInterface>> data_channels_;
std::vector<std::unique_ptr<MockDataChannelObserver>> data_observers_;
std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
std::vector<PeerConnectionInterface::IceConnectionState>
ice_connection_state_history_;
std::vector<PeerConnectionInterface::IceConnectionState>
standardized_ice_connection_state_history_;
std::vector<PeerConnectionInterface::PeerConnectionState>
peer_connection_state_history_;
std::vector<PeerConnectionInterface::IceGatheringState>
ice_gathering_state_history_;
std::vector<cricket::CandidatePairChangeEvent>
ice_candidate_pair_change_history_;
std::vector<PeerConnectionInterface::SignalingState>
peer_connection_signaling_state_history_;
webrtc::FakeRtcEventLogFactory* event_log_factory_;
// Number of ICE candidates expected. The default is no limit.
int candidates_expected_ = std::numeric_limits<int>::max();
// Variables for tracking delay stats on an audio track
int audio_packets_stat_ = 0;
double audio_delay_stat_ = 0.0;
uint64_t audio_samples_stat_ = 0;
uint64_t audio_concealed_stat_ = 0;
std::string rtp_stats_id_;
ScopedTaskSafety task_safety_;
friend class PeerConnectionIntegrationBaseTest;
};
class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
public:
virtual ~MockRtcEventLogOutput() = default;
MOCK_METHOD(bool, IsActive, (), (const, override));
MOCK_METHOD(bool, Write, (absl::string_view), (override));
};
// This helper object is used for both specifying how many audio/video frames
// are expected to be received for a caller/callee. It provides helper functions
// to specify these expectations. The object initially starts in a state of no
// expectations.
class MediaExpectations {
public:
enum ExpectFrames {
kExpectSomeFrames,
kExpectNoFrames,
kNoExpectation,
};
void ExpectBidirectionalAudioAndVideo() {
ExpectBidirectionalAudio();
ExpectBidirectionalVideo();
}
void ExpectBidirectionalAudio() {
CallerExpectsSomeAudio();
CalleeExpectsSomeAudio();
}
void ExpectNoAudio() {
CallerExpectsNoAudio();
CalleeExpectsNoAudio();
}
void ExpectBidirectionalVideo() {
CallerExpectsSomeVideo();
CalleeExpectsSomeVideo();
}
void ExpectNoVideo() {
CallerExpectsNoVideo();
CalleeExpectsNoVideo();
}
void CallerExpectsSomeAudioAndVideo() {
CallerExpectsSomeAudio();
CallerExpectsSomeVideo();
}
void CalleeExpectsSomeAudioAndVideo() {
CalleeExpectsSomeAudio();
CalleeExpectsSomeVideo();
}
// Caller's audio functions.
void CallerExpectsSomeAudio(
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
caller_audio_expectation_ = kExpectSomeFrames;
caller_audio_frames_expected_ = expected_audio_frames;
}
void CallerExpectsNoAudio() {
caller_audio_expectation_ = kExpectNoFrames;
caller_audio_frames_expected_ = 0;
}
// Caller's video functions.
void CallerExpectsSomeVideo(
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
caller_video_expectation_ = kExpectSomeFrames;
caller_video_frames_expected_ = expected_video_frames;
}
void CallerExpectsNoVideo() {
caller_video_expectation_ = kExpectNoFrames;
caller_video_frames_expected_ = 0;
}
// Callee's audio functions.
void CalleeExpectsSomeAudio(
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
callee_audio_expectation_ = kExpectSomeFrames;
callee_audio_frames_expected_ = expected_audio_frames;
}
void CalleeExpectsNoAudio() {
callee_audio_expectation_ = kExpectNoFrames;
callee_audio_frames_expected_ = 0;
}
// Callee's video functions.
void CalleeExpectsSomeVideo(
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
callee_video_expectation_ = kExpectSomeFrames;
callee_video_frames_expected_ = expected_video_frames;
}
void CalleeExpectsNoVideo() {
callee_video_expectation_ = kExpectNoFrames;
callee_video_frames_expected_ = 0;
}
ExpectFrames caller_audio_expectation_ = kNoExpectation;
ExpectFrames caller_video_expectation_ = kNoExpectation;
ExpectFrames callee_audio_expectation_ = kNoExpectation;
ExpectFrames callee_video_expectation_ = kNoExpectation;
int caller_audio_frames_expected_ = 0;
int caller_video_frames_expected_ = 0;
int callee_audio_frames_expected_ = 0;
int callee_video_frames_expected_ = 0;
};
class MockIceTransport : public webrtc::IceTransportInterface {
public:
MockIceTransport(const std::string& name, int component)
: internal_(std::make_unique<cricket::FakeIceTransport>(
name,
component,
nullptr /* network_thread */)) {}
~MockIceTransport() = default;
cricket::IceTransportInternal* internal() { return internal_.get(); }
private:
std::unique_ptr<cricket::FakeIceTransport> internal_;
};
class MockIceTransportFactory : public IceTransportFactory {
public:
~MockIceTransportFactory() override = default;
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
const std::string& transport_name,
int component,
IceTransportInit init) {
RecordIceTransportCreated();
return rtc::make_ref_counted<MockIceTransport>(transport_name, component);
}
MOCK_METHOD(void, RecordIceTransportCreated, ());
};
// Tests two PeerConnections connecting to each other end-to-end, using a
// virtual network, fake A/V capture and fake encoder/decoders. The
// PeerConnections share the threads/socket servers, but use separate versions
// of everything else (including "PeerConnectionFactory"s).
class PeerConnectionIntegrationBaseTest : public ::testing::Test {
public:
PeerConnectionIntegrationBaseTest(
SdpSemantics sdp_semantics,
absl::optional<std::string> field_trials = absl::nullopt)
: sdp_semantics_(sdp_semantics),
ss_(new rtc::VirtualSocketServer()),
fss_(new rtc::FirewallSocketServer(ss_.get())),
network_thread_(new rtc::Thread(fss_.get())),
worker_thread_(rtc::Thread::Create()),
// TODO(bugs.webrtc.org/10335): Pass optional ScopedKeyValueConfig.
field_trials_(new test::ScopedKeyValueConfig(
field_trials.has_value() ? *field_trials : "")) {
network_thread_->SetName("PCNetworkThread", this);
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
// The PeerConnections should be deleted before the TurnCustomizers.
// A TurnPort is created with a raw pointer to a TurnCustomizer. The
// TurnPort has the same lifetime as the PeerConnection, so it's expected
// that the TurnCustomizer outlives the life of the PeerConnection or else
// when Send() is called it will hit a seg fault.
if (caller_) {
caller_->set_signaling_message_receiver(nullptr);
caller_->pc()->Close();
delete SetCallerPcWrapperAndReturnCurrent(nullptr);
}
if (callee_) {
callee_->set_signaling_message_receiver(nullptr);
callee_->pc()->Close();
delete SetCalleePcWrapperAndReturnCurrent(nullptr);
}
// If turn servers were created for the test they need to be destroyed on
// the network thread.
SendTask(network_thread(), [this] {
turn_servers_.clear();
turn_customizers_.clear();
});
}
bool SignalingStateStable() {
return caller_->SignalingStateStable() && callee_->SignalingStateStable();
}
bool DtlsConnected() {
// TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
// are connected. This is an important distinction. Once we have separate
// ICE and DTLS state, this check needs to use the DTLS state.
return (callee()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
callee()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
(caller()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
caller()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted);
}
// When `event_log_factory` is null, the default implementation of the event
// log factory will be used.
std::unique_ptr<PeerConnectionIntegrationWrapper> CreatePeerConnectionWrapper(
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
bool create_media_engine = true) {
RTCConfiguration modified_config;
if (config) {
modified_config = *config;
}
modified_config.sdp_semantics = sdp_semantics_;
if (!dependencies.cert_generator) {
dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
}
std::unique_ptr<PeerConnectionIntegrationWrapper> client(
new PeerConnectionIntegrationWrapper(debug_name));
if (!client->Init(options, &modified_config, std::move(dependencies),
fss_.get(), network_thread_.get(), worker_thread_.get(),
std::move(event_log_factory), reset_encoder_factory,
reset_decoder_factory, create_media_engine)) {
return nullptr;
}
return client;
}
std::unique_ptr<PeerConnectionIntegrationWrapper>
CreatePeerConnectionWrapperWithFakeRtcEventLog(
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
return CreatePeerConnectionWrapper(
debug_name, options, config, std::move(dependencies),
std::make_unique<webrtc::FakeRtcEventLogFactory>(),
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
}
bool CreatePeerConnectionWrappers() {
return CreatePeerConnectionWrappersWithConfig(
PeerConnectionInterface::RTCConfiguration(),
PeerConnectionInterface::RTCConfiguration());
}
bool CreatePeerConnectionWrappersWithSdpSemantics(
SdpSemantics caller_semantics,
SdpSemantics callee_semantics) {
// Can't specify the sdp_semantics in the passed-in configuration since it
// will be overwritten by CreatePeerConnectionWrapper with whatever is
// stored in sdp_semantics_. So get around this by modifying the instance
// variable before calling CreatePeerConnectionWrapper for the caller and
// callee PeerConnections.
SdpSemantics original_semantics = sdp_semantics_;
sdp_semantics_ = caller_semantics;
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
sdp_semantics_ = callee_semantics;
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
sdp_semantics_ = original_semantics;
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfig(
const PeerConnectionInterface::RTCConfiguration& caller_config,
const PeerConnectionInterface::RTCConfiguration& callee_config) {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, &caller_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, &callee_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfigAndDeps(
const PeerConnectionInterface::RTCConfiguration& caller_config,
webrtc::PeerConnectionDependencies caller_dependencies,
const PeerConnectionInterface::RTCConfiguration& callee_config,
webrtc::PeerConnectionDependencies callee_dependencies) {
caller_ =
CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
std::move(caller_dependencies), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
callee_ =
CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
std::move(callee_dependencies), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithOptions(
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options) {
caller_ = CreatePeerConnectionWrapper(
"Caller", &caller_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
callee_ = CreatePeerConnectionWrapper(
"Callee", &callee_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
PeerConnectionInterface::RTCConfiguration default_config;
caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Caller", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Callee", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
return caller_ && callee_;
}
std::unique_ptr<PeerConnectionIntegrationWrapper>
CreatePeerConnectionWrapperWithAlternateKey() {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
cert_generator->use_alternate_key();
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
std::move(dependencies), nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
}
bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/!caller_to_callee,
/*reset_decoder_factory=*/caller_to_callee);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/caller_to_callee,
/*reset_decoder_factory=*/!caller_to_callee);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithoutMediaEngine() {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false,
/*create_media_engine=*/false);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false,
/*create_media_engine=*/false);
return caller_ && callee_;
}
cricket::TestTurnServer* CreateTurnServer(
rtc::SocketAddress internal_address,
rtc::SocketAddress external_address,
cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
const std::string& common_name = "test turn server") {
rtc::Thread* thread = network_thread();
rtc::SocketFactory* socket_factory = fss_.get();
std::unique_ptr<cricket::TestTurnServer> turn_server;
SendTask(network_thread(), [&] {
turn_server = std::make_unique<cricket::TestTurnServer>(
thread, socket_factory, internal_address, external_address, type,
/*ignore_bad_certs=*/true, common_name);
});
turn_servers_.push_back(std::move(turn_server));
// Interactions with the turn server should be done on the network thread.
return turn_servers_.back().get();
}
cricket::TestTurnCustomizer* CreateTurnCustomizer() {
std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer;
SendTask(network_thread(), [&] {
turn_customizer = std::make_unique<cricket::TestTurnCustomizer>();
});
turn_customizers_.push_back(std::move(turn_customizer));
// Interactions with the turn customizer should be done on the network
// thread.
return turn_customizers_.back().get();
}
// Checks that the function counters for a TestTurnCustomizer are greater than
// 0.
void ExpectTurnCustomizerCountersIncremented(
cricket::TestTurnCustomizer* turn_customizer) {
SendTask(network_thread(), [turn_customizer] {
EXPECT_GT(turn_customizer->allow_channel_data_cnt_, 0u);
EXPECT_GT(turn_customizer->modify_cnt_, 0u);
});
}
// Once called, SDP blobs and ICE candidates will be automatically signaled
// between PeerConnections.
void ConnectFakeSignaling() {
caller_->set_signaling_message_receiver(callee_.get());
callee_->set_signaling_message_receiver(caller_.get());
}
// Once called, SDP blobs will be automatically signaled between
// PeerConnections. Note that ICE candidates will not be signaled unless they
// are in the exchanged SDP blobs.
void ConnectFakeSignalingForSdpOnly() {
ConnectFakeSignaling();
SetSignalIceCandidates(false);
}
void SetSignalingDelayMs(int delay_ms) {
caller_->set_signaling_delay_ms(delay_ms);
callee_->set_signaling_delay_ms(delay_ms);
}
void SetSignalIceCandidates(bool signal) {
caller_->set_signal_ice_candidates(signal);
callee_->set_signal_ice_candidates(signal);
}
// Messages may get lost on the unreliable DataChannel, so we send multiple
// times to avoid test flakiness.
void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
const std::string& data,
int retries) {
for (int i = 0; i < retries; ++i) {
dc->Send(DataBuffer(data));
}
}
rtc::Thread* network_thread() { return network_thread_.get(); }
rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
PeerConnectionIntegrationWrapper* caller() { return caller_.get(); }
// Destroy peerconnections.
// This can be used to ensure that all pointers to on-stack mocks
// get dropped before exit.
void DestroyPeerConnections() {
if (caller_) {
caller_->pc()->Close();
}
if (callee_) {
callee_->pc()->Close();
}
caller_.reset();
callee_.reset();
}
// Set the `caller_` to the `wrapper` passed in and return the
// original `caller_`.
PeerConnectionIntegrationWrapper* SetCallerPcWrapperAndReturnCurrent(
PeerConnectionIntegrationWrapper* wrapper) {
PeerConnectionIntegrationWrapper* old = caller_.release();
caller_.reset(wrapper);
return old;
}
PeerConnectionIntegrationWrapper* callee() { return callee_.get(); }
// Set the `callee_` to the `wrapper` passed in and return the
// original `callee_`.
PeerConnectionIntegrationWrapper* SetCalleePcWrapperAndReturnCurrent(
PeerConnectionIntegrationWrapper* wrapper) {
PeerConnectionIntegrationWrapper* old = callee_.release();
callee_.reset(wrapper);
return old;
}
void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) {
SendTask(network_thread(), [this, caller_flags] {
caller()->port_allocator()->set_flags(caller_flags);
});
SendTask(network_thread(), [this, callee_flags] {
callee()->port_allocator()->set_flags(callee_flags);
});
}
rtc::FirewallSocketServer* firewall() const { return fss_.get(); }
// Expects the provided number of new frames to be received within
// kMaxWaitForFramesMs. The new expected frames are specified in
// `media_expectations`. Returns false if any of the expectations were
// not met.
bool ExpectNewFrames(const MediaExpectations& media_expectations) {
// Make sure there are no bogus tracks confusing the issue.
caller()->RemoveUnusedVideoRenderers();
callee()->RemoveUnusedVideoRenderers();
// First initialize the expected frame counts based upon the current
// frame count.
int total_caller_audio_frames_expected = caller()->audio_frames_received();
if (media_expectations.caller_audio_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_caller_audio_frames_expected +=
media_expectations.caller_audio_frames_expected_;
}
int total_caller_video_frames_expected =
caller()->min_video_frames_received_per_track();
if (media_expectations.caller_video_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_caller_video_frames_expected +=
media_expectations.caller_video_frames_expected_;
}
int total_callee_audio_frames_expected = callee()->audio_frames_received();
if (media_expectations.callee_audio_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_callee_audio_frames_expected +=
media_expectations.callee_audio_frames_expected_;
}
int total_callee_video_frames_expected =
callee()->min_video_frames_received_per_track();
if (media_expectations.callee_video_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_callee_video_frames_expected +=
media_expectations.callee_video_frames_expected_;
}
// Wait for the expected frames.
EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
total_caller_audio_frames_expected &&
caller()->min_video_frames_received_per_track() >=
total_caller_video_frames_expected &&
callee()->audio_frames_received() >=
total_callee_audio_frames_expected &&
callee()->min_video_frames_received_per_track() >=
total_callee_video_frames_expected,
kMaxWaitForFramesMs);
bool expectations_correct =
caller()->audio_frames_received() >=
total_caller_audio_frames_expected &&
caller()->min_video_frames_received_per_track() >=
total_caller_video_frames_expected &&
callee()->audio_frames_received() >=
total_callee_audio_frames_expected &&
callee()->min_video_frames_received_per_track() >=
total_callee_video_frames_expected;
// After the combined wait, print out a more detailed message upon
// failure.
EXPECT_GE(caller()->audio_frames_received(),
total_caller_audio_frames_expected);
EXPECT_GE(caller()->min_video_frames_received_per_track(),
total_caller_video_frames_expected);
EXPECT_GE(callee()->audio_frames_received(),
total_callee_audio_frames_expected);
EXPECT_GE(callee()->min_video_frames_received_per_track(),
total_callee_video_frames_expected);
// We want to make sure nothing unexpected was received.
if (media_expectations.caller_audio_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(caller()->audio_frames_received(),
total_caller_audio_frames_expected);
if (caller()->audio_frames_received() !=
total_caller_audio_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.caller_video_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(caller()->min_video_frames_received_per_track(),
total_caller_video_frames_expected);
if (caller()->min_video_frames_received_per_track() !=
total_caller_video_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.callee_audio_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(callee()->audio_frames_received(),
total_callee_audio_frames_expected);
if (callee()->audio_frames_received() !=
total_callee_audio_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.callee_video_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(callee()->min_video_frames_received_per_track(),
total_callee_video_frames_expected);
if (callee()->min_video_frames_received_per_track() !=
total_callee_video_frames_expected) {
expectations_correct = false;
}
}
return expectations_correct;
}
void ClosePeerConnections() {
if (caller())
caller()->pc()->Close();
if (callee())
callee()->pc()->Close();
}
void TestNegotiatedCipherSuite(
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options,
int expected_cipher_suite) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
callee_options));
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
bool remote_gcm_enabled,
bool aes_ctr_enabled,
int expected_cipher_suite) {
PeerConnectionFactory::Options caller_options;
caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
local_gcm_enabled;
caller_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher =
aes_ctr_enabled;
PeerConnectionFactory::Options callee_options;
callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
remote_gcm_enabled;
callee_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher =
aes_ctr_enabled;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
const FieldTrialsView& trials() const { return *field_trials_.get(); }
protected:
SdpSemantics sdp_semantics_;
private:
rtc::AutoThread main_thread_; // Used as the signal thread by most tests.
// `ss_` is used by `network_thread_` so it must be destroyed later.
std::unique_ptr<rtc::VirtualSocketServer> ss_;
std::unique_ptr<rtc::FirewallSocketServer> fss_;
// `network_thread_` and `worker_thread_` are used by both
// `caller_` and `callee_` so they must be destroyed
// later.
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<rtc::Thread> worker_thread_;
// The turn servers and turn customizers should be accessed & deleted on the
// network thread to avoid a race with the socket read/write that occurs
// on the network thread.
std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
std::unique_ptr<PeerConnectionIntegrationWrapper> caller_;
std::unique_ptr<PeerConnectionIntegrationWrapper> callee_;
std::unique_ptr<FieldTrialsView> field_trials_;
};
} // namespace webrtc
#endif // PC_TEST_INTEGRATION_TEST_HELPERS_H_
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