blob: b50489d5340f9aefc529c6f3c46443fd5ba6869e [file] [log] [blame]
/*
* Copyright 2018 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 <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/create_peerconnection_factory.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_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/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/port_interface.h"
#include "p2p/base/test_turn_server.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/frame_generator_capturer_video_track_source.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.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/location.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/test_certificate_verifier.h"
#include "rtc_base/thread.h"
#include "rtc_base/virtual_socket_server.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace {
static const int kDefaultTestTimeMs = 15000;
static const int kRampUpTimeMs = 5000;
static const int kPollIntervalTimeMs = 50;
static const int kDefaultTimeoutMs = 10000;
static const rtc::SocketAddress kDefaultLocalAddress("1.1.1.1", 0);
static const char kTurnInternalAddress[] = "88.88.88.0";
static const char kTurnExternalAddress[] = "88.88.88.1";
static const int kTurnInternalPort = 3478;
static const int kTurnExternalPort = 0;
// The video's configured max bitrate in webrtcvideoengine.cc is 1.7 Mbps.
// Setting the network bandwidth to 1 Mbps allows the video's bitrate to push
// the network's limitations.
static const int kNetworkBandwidth = 1000000;
} // namespace
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
// This is an end to end test to verify that BWE is functioning when setting
// up a one to one call at the PeerConnection level. The intention of the test
// is to catch potential regressions for different ICE path configurations. The
// test uses a VirtualSocketServer for it's underlying simulated network and
// fake audio and video sources. The test is based upon rampup_tests.cc, but
// instead is at the PeerConnection level and uses a different fake network
// (rampup_tests.cc uses SimulatedNetwork). In the future, this test could
// potentially test different network conditions and test video quality as well
// (video_quality_test.cc does this, but at the call level).
//
// The perf test results are printed using the perf test support. If the
// isolated_script_test_perf_output flag is specified in test_main.cc, then
// the results are written to a JSON formatted file for the Chrome perf
// dashboard. Since this test is a webrtc_perf_test, it will be run in the perf
// console every webrtc commit.
class PeerConnectionWrapperForRampUpTest : public PeerConnectionWrapper {
public:
using PeerConnectionWrapper::PeerConnectionWrapper;
PeerConnectionWrapperForRampUpTest(
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
rtc::scoped_refptr<PeerConnectionInterface> pc,
std::unique_ptr<MockPeerConnectionObserver> observer)
: PeerConnectionWrapper::PeerConnectionWrapper(pc_factory,
pc,
std::move(observer)) {}
bool AddIceCandidates(std::vector<const IceCandidateInterface*> candidates) {
bool success = true;
for (const auto candidate : candidates) {
if (!pc()->AddIceCandidate(candidate)) {
success = false;
}
}
return success;
}
rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack(
FrameGeneratorCapturerVideoTrackSource::Config config,
Clock* clock) {
video_track_sources_.emplace_back(
new rtc::RefCountedObject<FrameGeneratorCapturerVideoTrackSource>(
config, clock, /*is_screencast=*/false));
video_track_sources_.back()->Start();
return rtc::scoped_refptr<VideoTrackInterface>(
pc_factory()->CreateVideoTrack(rtc::CreateRandomUuid(),
video_track_sources_.back()));
}
rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack(
const cricket::AudioOptions options) {
rtc::scoped_refptr<AudioSourceInterface> source =
pc_factory()->CreateAudioSource(options);
return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(), source);
}
private:
std::vector<rtc::scoped_refptr<FrameGeneratorCapturerVideoTrackSource>>
video_track_sources_;
};
// TODO(shampson): Paramaterize the test to run for both Plan B & Unified Plan.
class PeerConnectionRampUpTest : public ::testing::Test {
public:
PeerConnectionRampUpTest()
: clock_(Clock::GetRealTimeClock()),
virtual_socket_server_(new rtc::VirtualSocketServer()),
firewall_socket_server_(
new rtc::FirewallSocketServer(virtual_socket_server_.get())),
network_thread_(new rtc::Thread(firewall_socket_server_.get())),
worker_thread_(rtc::Thread::Create()) {
network_thread_->SetName("PCNetworkThread", this);
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
virtual_socket_server_->set_bandwidth(kNetworkBandwidth / 8);
pc_factory_ = CreatePeerConnectionFactory(
network_thread_.get(), worker_thread_.get(), rtc::Thread::Current(),
rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */, nullptr /* audio_processing */);
}
virtual ~PeerConnectionRampUpTest() {
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { turn_servers_.clear(); });
}
bool CreatePeerConnectionWrappers(const RTCConfiguration& caller_config,
const RTCConfiguration& callee_config) {
caller_ = CreatePeerConnectionWrapper(caller_config);
callee_ = CreatePeerConnectionWrapper(callee_config);
return caller_ && callee_;
}
std::unique_ptr<PeerConnectionWrapperForRampUpTest>
CreatePeerConnectionWrapper(const RTCConfiguration& config) {
auto* fake_network_manager = new rtc::FakeNetworkManager();
fake_network_manager->AddInterface(kDefaultLocalAddress);
fake_network_managers_.emplace_back(fake_network_manager);
auto observer = std::make_unique<MockPeerConnectionObserver>();
webrtc::PeerConnectionDependencies dependencies(observer.get());
cricket::BasicPortAllocator* port_allocator =
new cricket::BasicPortAllocator(fake_network_manager);
port_allocator->set_step_delay(cricket::kDefaultStepDelay);
dependencies.allocator =
std::unique_ptr<cricket::BasicPortAllocator>(port_allocator);
dependencies.tls_cert_verifier =
std::make_unique<rtc::TestCertificateVerifier>();
auto pc =
pc_factory_->CreatePeerConnection(config, std::move(dependencies));
if (!pc) {
return nullptr;
}
return std::make_unique<PeerConnectionWrapperForRampUpTest>(
pc_factory_, pc, std::move(observer));
}
void SetupOneWayCall() {
ASSERT_TRUE(caller_);
ASSERT_TRUE(callee_);
FrameGeneratorCapturerVideoTrackSource::Config config;
caller_->AddTrack(caller_->CreateLocalVideoTrack(config, clock_));
// Disable highpass filter so that we can get all the test audio frames.
cricket::AudioOptions options;
options.highpass_filter = false;
caller_->AddTrack(caller_->CreateLocalAudioTrack(options));
// Do the SDP negotiation, and also exchange ice candidates.
ASSERT_TRUE(caller_->ExchangeOfferAnswerWith(callee_.get()));
ASSERT_TRUE_WAIT(
caller_->signaling_state() == PeerConnectionInterface::kStable,
kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(caller_->IsIceGatheringDone(), kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(callee_->IsIceGatheringDone(), kDefaultTimeoutMs);
// Connect an ICE candidate pairs.
ASSERT_TRUE(
callee_->AddIceCandidates(caller_->observer()->GetAllCandidates()));
ASSERT_TRUE(
caller_->AddIceCandidates(callee_->observer()->GetAllCandidates()));
// This means that ICE and DTLS are connected.
ASSERT_TRUE_WAIT(callee_->IsIceConnected(), kDefaultTimeoutMs);
ASSERT_TRUE_WAIT(caller_->IsIceConnected(), kDefaultTimeoutMs);
}
void CreateTurnServer(cricket::ProtocolType type,
const std::string& common_name = "test turn server") {
rtc::Thread* thread = network_thread();
std::unique_ptr<cricket::TestTurnServer> turn_server =
network_thread_->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
RTC_FROM_HERE, [thread, type, common_name] {
static const rtc::SocketAddress turn_server_internal_address{
kTurnInternalAddress, kTurnInternalPort};
static const rtc::SocketAddress turn_server_external_address{
kTurnExternalAddress, kTurnExternalPort};
return std::make_unique<cricket::TestTurnServer>(
thread, turn_server_internal_address,
turn_server_external_address, type,
true /*ignore_bad_certs=*/, common_name);
});
turn_servers_.push_back(std::move(turn_server));
}
// First runs the call for kRampUpTimeMs to ramp up the bandwidth estimate.
// Then runs the test for the remaining test time, grabbing the bandwidth
// estimation stat, every kPollIntervalTimeMs. When finished, averages the
// bandwidth estimations and prints the bandwidth estimation result as a perf
// metric.
void RunTest(const std::string& test_string) {
rtc::Thread::Current()->ProcessMessages(kRampUpTimeMs);
int number_of_polls =
(kDefaultTestTimeMs - kRampUpTimeMs) / kPollIntervalTimeMs;
int total_bwe = 0;
for (int i = 0; i < number_of_polls; ++i) {
rtc::Thread::Current()->ProcessMessages(kPollIntervalTimeMs);
total_bwe += static_cast<int>(GetCallerAvailableBitrateEstimate());
}
double average_bandwidth_estimate = total_bwe / number_of_polls;
std::string value_description =
"bwe_after_" + std::to_string(kDefaultTestTimeMs / 1000) + "_seconds";
test::PrintResult("peerconnection_ramp_up_", test_string, value_description,
average_bandwidth_estimate, "bwe", false);
}
rtc::Thread* network_thread() { return network_thread_.get(); }
rtc::FirewallSocketServer* firewall_socket_server() {
return firewall_socket_server_.get();
}
PeerConnectionWrapperForRampUpTest* caller() { return caller_.get(); }
PeerConnectionWrapperForRampUpTest* callee() { return callee_.get(); }
private:
// Gets the caller's outgoing available bitrate from the stats. Returns 0 if
// something went wrong. It takes the outgoing bitrate from the current
// selected ICE candidate pair's stats.
double GetCallerAvailableBitrateEstimate() {
auto stats = caller_->GetStats();
auto transport_stats = stats->GetStatsOfType<RTCTransportStats>();
if (transport_stats.size() == 0u ||
!transport_stats[0]->selected_candidate_pair_id.is_defined()) {
return 0;
}
std::string selected_ice_id =
transport_stats[0]->selected_candidate_pair_id.ValueToString();
// Use the selected ICE candidate pair ID to get the appropriate ICE stats.
const RTCIceCandidatePairStats ice_candidate_pair_stats =
stats->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>();
if (ice_candidate_pair_stats.available_outgoing_bitrate.is_defined()) {
return *ice_candidate_pair_stats.available_outgoing_bitrate;
}
// We couldn't get the |available_outgoing_bitrate| for the active candidate
// pair.
return 0;
}
Clock* const clock_;
// The turn servers should be accessed & deleted on the network thread to
// avoid a race with the socket read/write which occurs on the network thread.
std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
// |virtual_socket_server_| is used by |network_thread_| so it must be
// destroyed later.
// TODO(bugs.webrtc.org/7668): We would like to update the virtual network we
// use for this test. VirtualSocketServer isn't ideal because:
// 1) It uses the same queue & network capacity for both directions.
// 2) VirtualSocketServer implements how the network bandwidth affects the
// send delay differently than the SimulatedNetwork, used by the
// FakeNetworkPipe. It would be ideal if all of levels of virtual
// networks used in testing were consistent.
// We would also like to update this test to record the time to ramp up,
// down, and back up (similar to in rampup_tests.cc). This is problematic with
// the VirtualSocketServer. The first ramp down time is very noisy and the
// second ramp up time can take up to 300 seconds, most likely due to a built
// up queue.
std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
std::unique_ptr<rtc::FirewallSocketServer> firewall_socket_server_;
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<rtc::Thread> worker_thread_;
// The |pc_factory| uses |network_thread_| & |worker_thread_|, so it must be
// destroyed first.
std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_network_managers_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
std::unique_ptr<PeerConnectionWrapperForRampUpTest> caller_;
std::unique_ptr<PeerConnectionWrapperForRampUpTest> callee_;
};
TEST_F(PeerConnectionRampUpTest, TurnOverTCP) {
CreateTurnServer(cricket::ProtocolType::PROTO_TCP);
PeerConnectionInterface::IceServer ice_server;
std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
":" + std::to_string(kTurnInternalPort) +
"?transport=tcp";
ice_server.urls.push_back(ice_server_url);
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
client_2_config.type = PeerConnectionInterface::kRelay;
ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
SetupOneWayCall();
RunTest("turn_over_tcp");
}
TEST_F(PeerConnectionRampUpTest, TurnOverUDP) {
CreateTurnServer(cricket::ProtocolType::PROTO_UDP);
PeerConnectionInterface::IceServer ice_server;
std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
":" + std::to_string(kTurnInternalPort);
ice_server.urls.push_back(ice_server_url);
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
client_2_config.type = PeerConnectionInterface::kRelay;
ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
SetupOneWayCall();
RunTest("turn_over_udp");
}
TEST_F(PeerConnectionRampUpTest, TurnOverTLS) {
CreateTurnServer(cricket::ProtocolType::PROTO_TLS, kTurnInternalAddress);
PeerConnectionInterface::IceServer ice_server;
std::string ice_server_url = "turns:" + std::string(kTurnInternalAddress) +
":" + std::to_string(kTurnInternalPort) +
"?transport=tcp";
ice_server.urls.push_back(ice_server_url);
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
client_2_config.type = PeerConnectionInterface::kRelay;
ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
SetupOneWayCall();
RunTest("turn_over_tls");
}
TEST_F(PeerConnectionRampUpTest, UDPPeerToPeer) {
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.tcp_candidate_policy =
PeerConnection::kTcpCandidatePolicyDisabled;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.tcp_candidate_policy =
PeerConnection::kTcpCandidatePolicyDisabled;
ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
SetupOneWayCall();
RunTest("udp_peer_to_peer");
}
TEST_F(PeerConnectionRampUpTest, TCPPeerToPeer) {
firewall_socket_server()->set_udp_sockets_enabled(false);
ASSERT_TRUE(CreatePeerConnectionWrappers(
PeerConnectionInterface::RTCConfiguration(),
PeerConnectionInterface::RTCConfiguration()));
SetupOneWayCall();
RunTest("tcp_peer_to_peer");
}
} // namespace webrtc