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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.
*/
#include "pc/peer_connection_factory.h"
#include <stddef.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#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/create_peerconnection_factory.h"
#include "api/data_channel_interface.h"
#include "api/jsep.h"
#include "api/media_stream_interface.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 "media/base/fake_frame_source.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/port.h"
#include "p2p/base/port_interface.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_video_track_source.h"
#include "rtc_base/socket_address.h"
#include "test/gtest.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/fake_video_track_renderer.h"
using webrtc::DataChannelInterface;
using webrtc::FakeVideoTrackRenderer;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::VideoTrackInterface;
using webrtc::VideoTrackSourceInterface;
namespace {
static const char kStunIceServer[] = "stun:stun.l.google.com:19302";
static const char kTurnIceServer[] = "turn:test.com:1234";
static const char kTurnIceServerWithTransport[] =
"turn:hello.com?transport=tcp";
static const char kSecureTurnIceServer[] = "turns:hello.com?transport=tcp";
static const char kSecureTurnIceServerWithoutTransportParam[] =
"turns:hello.com:443";
static const char kSecureTurnIceServerWithoutTransportAndPortParam[] =
"turns:hello.com";
static const char kTurnIceServerWithNoUsernameInUri[] = "turn:test.com:1234";
static const char kTurnPassword[] = "turnpassword";
static const int kDefaultStunPort = 3478;
static const int kDefaultStunTlsPort = 5349;
static const char kTurnUsername[] = "test";
static const char kStunIceServerWithIPv4Address[] = "stun:1.2.3.4:1234";
static const char kStunIceServerWithIPv4AddressWithoutPort[] = "stun:1.2.3.4";
static const char kStunIceServerWithIPv6Address[] = "stun:[2401:fa00:4::]:1234";
static const char kStunIceServerWithIPv6AddressWithoutPort[] =
"stun:[2401:fa00:4::]";
static const char kTurnIceServerWithIPv6Address[] = "turn:[2401:fa00:4::]:1234";
class NullPeerConnectionObserver : public PeerConnectionObserver {
public:
virtual ~NullPeerConnectionObserver() = default;
void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) override {}
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {}
void OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> stream) override {}
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) override {}
void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) override {}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
}
};
} // namespace
class PeerConnectionFactoryTest : public ::testing::Test {
void SetUp() {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
// Use fake audio device module since we're only testing the interface
// level, and using a real one could make tests flaky e.g. when run in
// parallel.
factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
rtc::scoped_refptr<webrtc::AudioDeviceModule>(
FakeAudioCaptureModule::Create()),
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */);
ASSERT_TRUE(factory_.get() != NULL);
port_allocator_.reset(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
raw_port_allocator_ = port_allocator_.get();
}
protected:
void VerifyStunServers(cricket::ServerAddresses stun_servers) {
EXPECT_EQ(stun_servers, raw_port_allocator_->stun_servers());
}
void VerifyTurnServers(std::vector<cricket::RelayServerConfig> turn_servers) {
EXPECT_EQ(turn_servers.size(), raw_port_allocator_->turn_servers().size());
for (size_t i = 0; i < turn_servers.size(); ++i) {
ASSERT_EQ(1u, turn_servers[i].ports.size());
EXPECT_EQ(1u, raw_port_allocator_->turn_servers()[i].ports.size());
EXPECT_EQ(
turn_servers[i].ports[0].address.ToString(),
raw_port_allocator_->turn_servers()[i].ports[0].address.ToString());
EXPECT_EQ(turn_servers[i].ports[0].proto,
raw_port_allocator_->turn_servers()[i].ports[0].proto);
EXPECT_EQ(turn_servers[i].credentials.username,
raw_port_allocator_->turn_servers()[i].credentials.username);
EXPECT_EQ(turn_servers[i].credentials.password,
raw_port_allocator_->turn_servers()[i].credentials.password);
}
}
void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) {
EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
EXPECT_GT(codec.num_channels, 0);
}
void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec) {
EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
}
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
NullPeerConnectionObserver observer_;
std::unique_ptr<cricket::FakePortAllocator> port_allocator_;
// Since the PC owns the port allocator after it's been initialized,
// this should only be used when known to be safe.
cricket::FakePortAllocator* raw_port_allocator_;
};
// Verify creation of PeerConnection using internal ADM, video factory and
// internal libjingle threads.
// TODO(henrika): disabling this test since relying on real audio can result in
// flaky tests and focus on details that are out of scope for you might expect
// for a PeerConnectionFactory unit test.
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details.
TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
webrtc::CreatePeerConnectionFactory(
nullptr /* network_thread */, nullptr /* worker_thread */,
nullptr /* signaling_thread */, nullptr /* default_adm */,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */));
NullPeerConnectionObserver observer;
webrtc::PeerConnectionInterface::RTCConfiguration config;
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
config, nullptr, std::move(cert_generator), &observer));
EXPECT_TRUE(pc.get() != nullptr);
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
webrtc::RtpCapabilities audio_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
VerifyAudioCodecCapability(codec);
}
EXPECT_FALSE(audio_capabilities.header_extensions.empty());
for (const auto& header_extension : audio_capabilities.header_extensions) {
EXPECT_FALSE(header_extension.uri.empty());
}
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
webrtc::RtpCapabilities video_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
VerifyVideoCodecCapability(codec);
}
EXPECT_FALSE(video_capabilities.header_extensions.empty());
for (const auto& header_extension : video_capabilities.header_extensions) {
EXPECT_FALSE(header_extension.uri.empty());
}
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
webrtc::RtpCapabilities data_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
webrtc::RtpCapabilities audio_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
VerifyAudioCodecCapability(codec);
}
EXPECT_FALSE(audio_capabilities.header_extensions.empty());
for (const auto& header_extension : audio_capabilities.header_extensions) {
EXPECT_FALSE(header_extension.uri.empty());
}
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
webrtc::RtpCapabilities video_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
VerifyVideoCodecCapability(codec);
}
EXPECT_FALSE(video_capabilities.header_extensions.empty());
for (const auto& header_extension : video_capabilities.header_extensions) {
EXPECT_FALSE(header_extension.uri.empty());
}
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
webrtc::RtpCapabilities data_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
}
// This test verifies creation of PeerConnection with valid STUN and TURN
// configuration. Also verifies the URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServer;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServer;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServerWithTransport;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("stun.l.google.com", 19302);
stun_servers.insert(stun1);
VerifyStunServers(stun_servers);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername,
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn1);
cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername,
kTurnPassword, cricket::PROTO_TCP);
turn_servers.push_back(turn2);
VerifyTurnServers(turn_servers);
}
// This test verifies creation of PeerConnection with valid STUN and TURN
// configuration. Also verifies the list of URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back(kStunIceServer);
ice_server.urls.push_back(kTurnIceServer);
ice_server.urls.push_back(kTurnIceServerWithTransport);
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("stun.l.google.com", 19302);
stun_servers.insert(stun1);
VerifyStunServers(stun_servers);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername,
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn1);
cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername,
kTurnPassword, cricket::PROTO_TCP);
turn_servers.push_back(turn2);
VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServer;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServerWithNoUsernameInUri;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn("test.com", 1234, kTurnUsername,
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn);
VerifyTurnServers(turn_servers);
}
// This test verifies the PeerConnection created properly with TURN url which
// has transport parameter in it.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kTurnIceServerWithTransport;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn("hello.com", kDefaultStunPort, kTurnUsername,
kTurnPassword, cricket::PROTO_TCP);
turn_servers.push_back(turn);
VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kSecureTurnIceServer;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kSecureTurnIceServerWithoutTransportParam;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kSecureTurnIceServerWithoutTransportAndPortParam;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn1("hello.com", kDefaultStunTlsPort,
kTurnUsername, kTurnPassword,
cricket::PROTO_TLS);
turn_servers.push_back(turn1);
// TURNS with transport param should be default to tcp.
cricket::RelayServerConfig turn2("hello.com", 443, kTurnUsername,
kTurnPassword, cricket::PROTO_TLS);
turn_servers.push_back(turn2);
cricket::RelayServerConfig turn3("hello.com", kDefaultStunTlsPort,
kTurnUsername, kTurnPassword,
cricket::PROTO_TLS);
turn_servers.push_back(turn3);
VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServerWithIPv4Address;
config.servers.push_back(ice_server);
ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort;
config.servers.push_back(ice_server);
ice_server.uri = kStunIceServerWithIPv6Address;
config.servers.push_back(ice_server);
ice_server.uri = kStunIceServerWithIPv6AddressWithoutPort;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServerWithIPv6Address;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(
factory_->CreatePeerConnection(config, std::move(port_allocator_),
std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("1.2.3.4", 1234);
stun_servers.insert(stun1);
rtc::SocketAddress stun2("1.2.3.4", 3478);
stun_servers.insert(stun2); // Default port
rtc::SocketAddress stun3("2401:fa00:4::", 1234);
stun_servers.insert(stun3);
rtc::SocketAddress stun4("2401:fa00:4::", 3478);
stun_servers.insert(stun4); // Default port
VerifyStunServers(stun_servers);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn1("2401:fa00:4::", 1234, kTurnUsername,
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn1);
VerifyTurnServers(turn_servers);
}
// This test verifies the captured stream is rendered locally using a
// local video track.
TEST_F(PeerConnectionFactoryTest, LocalRendering) {
rtc::scoped_refptr<webrtc::FakeVideoTrackSource> source =
webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false);
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
ASSERT_TRUE(source.get() != NULL);
rtc::scoped_refptr<VideoTrackInterface> track(
factory_->CreateVideoTrack("testlabel", source));
ASSERT_TRUE(track.get() != NULL);
FakeVideoTrackRenderer local_renderer(track);
EXPECT_EQ(0, local_renderer.num_rendered_frames());
source->InjectFrame(frame_source.GetFrame());
EXPECT_EQ(1, local_renderer.num_rendered_frames());
EXPECT_FALSE(local_renderer.black_frame());
track->set_enabled(false);
source->InjectFrame(frame_source.GetFrame());
EXPECT_EQ(2, local_renderer.num_rendered_frames());
EXPECT_TRUE(local_renderer.black_frame());
track->set_enabled(true);
source->InjectFrame(frame_source.GetFrame());
EXPECT_EQ(3, local_renderer.num_rendered_frames());
EXPECT_FALSE(local_renderer.black_frame());
}
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