webrtc_m130/pc/peerconnectionfactory_unittest.cc
henrika 919dc2e843 Removes fallback from Linux PulseAudio to ALSA.
Unit test now checks that ADM:Init() works before any test runs.
It means that all tests will be skipped on bots that lack Pulse
support which is as how it worked before this CL as well. But then,
we detected the lack of support by checking that the audio layer had
changed from Pulse to Alsa.

As a consequence, I also decided to inject fake/mock ADMs in more
unit tests. One was actually already injected for other reasons
(see https://codereview.webrtc.org/2997383002/) but it had accidentally
been "reverted" later in combination with other changes.

To summarize: before this change we had a set of unit tests where real
audio was tested but it was not the intention of the test or required.
In addition, some Linux bots (VM:s) did not support PulseAudio and on
them the tests relied on a fallback mechanism to ALSA to work, i.e.,
we had a rather complex dependency on hardware. This dependency has now
been removed and it should result in more stable tests.

Bug: webrtc:7306, webrtc:7806
Change-Id: Ia0485658c04a4ef3b3f2dc0d557d73738067304b
Reviewed-on: https://webrtc-review.googlesource.com/8640
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20307}
2017-10-16 11:49:37 +00:00

364 lines
16 KiB
C++

/*
* 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 <memory>
#include <string>
#include <utility>
#include "api/mediastreaminterface.h"
#include "media/base/fakevideocapturer.h"
#include "p2p/base/fakeportallocator.h"
#include "pc/peerconnectionfactory.h"
#include "pc/test/fakeaudiocapturemodule.h"
#include "rtc_base/gunit.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
#include "pc/test/fakertccertificategenerator.h"
#include "pc/test/fakevideotrackrenderer.h"
using webrtc::DataChannelInterface;
using webrtc::FakeVideoTrackRenderer;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::VideoTrackSourceInterface;
using webrtc::VideoTrackInterface;
namespace {
static const char kStunIceServer[] = "stun:stun.l.google.com:19302";
static const char kTurnIceServer[] = "turn:test%40hello.com@test.com:1234";
static const char kTurnIceServerWithTransport[] =
"turn:test@hello.com?transport=tcp";
static const char kSecureTurnIceServer[] =
"turns:test@hello.com?transport=tcp";
static const char kSecureTurnIceServerWithoutTransportParam[] =
"turns:test_no_transport@hello.com:443";
static const char kSecureTurnIceServerWithoutTransportAndPortParam[] =
"turns:test_no_transport@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:test@[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(),
FakeAudioCaptureModule::Create(), nullptr, nullptr);
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);
}
}
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());
NullPeerConnectionObserver observer;
webrtc::PeerConnectionInterface::RTCConfiguration config;
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
config, nullptr, nullptr, std::move(cert_generator), &observer));
EXPECT_TRUE(pc.get() != nullptr);
}
// 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.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServerWithTransport;
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, nullptr, 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, "test@hello.com",
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn1);
cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, "test",
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.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
config, nullptr, 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, "test@hello.com",
kTurnPassword, cricket::PROTO_UDP);
turn_servers.push_back(turn1);
cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, "test",
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, nullptr, 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.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
config, nullptr, 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, "test",
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.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kSecureTurnIceServerWithoutTransportParam;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
ice_server.uri = kSecureTurnIceServerWithoutTransportAndPortParam;
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, nullptr, 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, "test",
kTurnPassword, cricket::PROTO_TLS);
turn_servers.push_back(turn1);
// TURNS with transport param should be default to tcp.
cricket::RelayServerConfig turn2("hello.com", 443, "test_no_transport",
kTurnPassword, cricket::PROTO_TLS);
turn_servers.push_back(turn2);
cricket::RelayServerConfig turn3("hello.com", kDefaultStunTlsPort,
"test_no_transport", 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.password = kTurnPassword;
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
config, nullptr, 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, "test", 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) {
cricket::FakeVideoCapturer* capturer = new cricket::FakeVideoCapturer();
// The source takes ownership of |capturer|, but we keep a raw pointer to
// inject fake frames.
rtc::scoped_refptr<VideoTrackSourceInterface> source(
factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(capturer), NULL));
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());
EXPECT_TRUE(capturer->CaptureFrame());
EXPECT_EQ(1, local_renderer.num_rendered_frames());
EXPECT_FALSE(local_renderer.black_frame());
track->set_enabled(false);
EXPECT_TRUE(capturer->CaptureFrame());
EXPECT_EQ(2, local_renderer.num_rendered_frames());
EXPECT_TRUE(local_renderer.black_frame());
track->set_enabled(true);
EXPECT_TRUE(capturer->CaptureFrame());
EXPECT_EQ(3, local_renderer.num_rendered_frames());
EXPECT_FALSE(local_renderer.black_frame());
}