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/*
* Copyright (c) 2022 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.
*/
// This file contains tests that verify that field trials do what they're
// supposed to do.
#include <memory>
#include <set>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/enable_media_with_defaults.h"
#include "api/environment/environment_factory.h"
#include "api/field_trials.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/internal/default_socket_server.h"
#include "rtc_base/socket_server.h"
#include "rtc_base/thread.h"
#include "system_wrappers/include/clock.h"
#include "test/create_test_field_trials.h"
#include "test/gtest.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
class PeerConnectionFieldTrialTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapper> WrapperPtr;
PeerConnectionFieldTrialTest()
: clock_(Clock::GetRealTimeClock()),
socket_server_(CreateDefaultSocketServer()),
main_thread_(socket_server_.get()) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
config_.servers.push_back(ice_server);
config_.sdp_semantics = SdpSemantics::kUnifiedPlan;
}
void TearDown() override { pc_factory_ = nullptr; }
void CreatePCFactory(absl::string_view field_trials) {
PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.signaling_thread = Thread::Current();
pcf_deps.env = CreateEnvironment(
std::make_unique<FieldTrials>(CreateTestFieldTrials(field_trials)));
pcf_deps.adm = FakeAudioCaptureModule::Create();
EnableMediaWithDefaults(pcf_deps);
pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
// Allow ADAPTER_TYPE_LOOPBACK to create PeerConnections with loopback in
// this test.
RTC_DCHECK(pc_factory_);
PeerConnectionFactoryInterface::Options options;
options.network_ignore_mask = 0;
pc_factory_->SetOptions(options);
}
WrapperPtr CreatePeerConnection() {
auto observer = std::make_unique<MockPeerConnectionObserver>();
auto result = pc_factory_->CreatePeerConnectionOrError(
config_, PeerConnectionDependencies(observer.get()));
RTC_CHECK(result.ok());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
Clock* const clock_;
std::unique_ptr<SocketServer> socket_server_;
AutoSocketServerThread main_thread_;
scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
PeerConnectionInterface::RTCConfiguration config_;
};
// Tests for the dependency descriptor field trial. The dependency descriptor
// field trial is implemented in media/engine/webrtc_video_engine.cc.
TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) {
CreatePCFactory("WebRTC-DependencyDescriptorAdvertised/Enabled/");
WrapperPtr caller = CreatePeerConnection();
caller->AddTransceiver(MediaType::VIDEO);
auto offer = caller->CreateOffer();
auto contents1 = offer->description()->contents();
ASSERT_EQ(1u, contents1.size());
const MediaContentDescription* media_description1 =
contents1[0].media_description();
EXPECT_EQ(MediaType::VIDEO, media_description1->type());
const RtpHeaderExtensions& rtp_header_extensions1 =
media_description1->rtp_header_extensions();
bool found =
absl::c_find_if(
rtp_header_extensions1, [](const RtpExtension& rtp_extension) {
return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
EXPECT_TRUE(found);
}
// Tests that dependency descriptor RTP header extensions can be exchanged
// via SDP munging, even if dependency descriptor field trial is disabled.
#ifdef WEBRTC_WIN
// TODO: crbug.com/webrtc/15876 - Test is flaky on Windows machines.
#define MAYBE_InjectDependencyDescriptor DISABLED_InjectDependencyDescriptor
#else
#define MAYBE_InjectDependencyDescriptor InjectDependencyDescriptor
#endif
TEST_F(PeerConnectionFieldTrialTest, MAYBE_InjectDependencyDescriptor) {
CreatePCFactory("WebRTC-DependencyDescriptorAdvertised/Disabled/");
WrapperPtr caller = CreatePeerConnection();
WrapperPtr callee = CreatePeerConnection();
caller->AddTransceiver(MediaType::VIDEO);
auto offer = caller->CreateOffer();
ContentInfos& contents1 = offer->description()->contents();
ASSERT_EQ(1u, contents1.size());
MediaContentDescription* media_description1 =
contents1[0].media_description();
EXPECT_EQ(MediaType::VIDEO, media_description1->type());
RtpHeaderExtensions rtp_header_extensions1 =
media_description1->rtp_header_extensions();
bool found1 =
absl::c_find_if(
rtp_header_extensions1, [](const RtpExtension& rtp_extension) {
return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
EXPECT_FALSE(found1);
std::set<int> existing_ids;
for (const RtpExtension& rtp_extension : rtp_header_extensions1) {
existing_ids.insert(rtp_extension.id);
}
// Find the currently unused RTP header extension ID.
int insert_id = 1;
std::set<int>::const_iterator iter = existing_ids.begin();
while (true) {
if (iter == existing_ids.end()) {
break;
}
if (*iter != insert_id) {
break;
}
insert_id++;
iter++;
}
rtp_header_extensions1.emplace_back(RtpExtension::kDependencyDescriptorUri,
insert_id);
media_description1->set_rtp_header_extensions(rtp_header_extensions1);
caller->SetLocalDescription(offer->Clone());
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
ContentInfos& contents2 = answer->description()->contents();
ASSERT_EQ(1u, contents2.size());
MediaContentDescription* media_description2 =
contents2[0].media_description();
EXPECT_EQ(MediaType::VIDEO, media_description2->type());
RtpHeaderExtensions rtp_header_extensions2 =
media_description2->rtp_header_extensions();
bool found2 =
absl::c_find_if(
rtp_header_extensions2, [](const RtpExtension& rtp_extension) {
return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions2.end();
EXPECT_TRUE(found2);
}
} // namespace webrtc
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