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/*
* Copyright 2017 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 <utility>
#include <vector>
#include "api/audio/audio_mixer.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/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_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 "pc/peer_connection_wrapper.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/thread.h"
#include "system_wrappers/include/metrics.h"
#include "test/gtest.h"
// This file contains unit tests that relate to the behavior of the
// SdpOfferAnswer module.
// Tests are writen as integration tests with PeerConnection, since the
// behaviors are still linked so closely that it is hard to test them in
// isolation.
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
namespace {
std::unique_ptr<rtc::Thread> CreateAndStartThread() {
auto thread = rtc::Thread::Create();
thread->Start();
return thread;
}
} // namespace
class SdpOfferAnswerTest : public ::testing::Test {
public:
SdpOfferAnswerTest()
// Note: We use a PeerConnectionFactory with a distinct
// signaling thread, so that thread handling can be tested.
: signaling_thread_(CreateAndStartThread()),
pc_factory_(
CreatePeerConnectionFactory(nullptr,
nullptr,
signaling_thread_.get(),
FakeAudioCaptureModule::Create(),
CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(),
CreateBuiltinVideoEncoderFactory(),
CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */,
nullptr /* audio_processing */)) {
webrtc::metrics::Reset();
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
return CreatePeerConnection(config);
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
const RTCConfiguration& config) {
auto observer = std::make_unique<MockPeerConnectionObserver>();
auto result = pc_factory_->CreatePeerConnectionOrError(
config, PeerConnectionDependencies(observer.get()));
EXPECT_TRUE(result.ok());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
protected:
std::unique_ptr<rtc::Thread> signaling_thread_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
private:
rtc::AutoThread main_thread_;
};
TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
// Verify that caller->observer->OnTrack() has been called with a
// proxied transceiver object.
ASSERT_EQ(callee->observer()->on_track_transceivers_.size(), 1u);
auto transceiver = callee->observer()->on_track_transceivers_[0];
// Since the signaling thread is not the current thread,
// this will DCHECK if the transceiver is not proxied.
transceiver->stopped();
}
} // namespace webrtc
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