blob: 8cb8415f33ff2927e823eae716aaf2cfc4c42108 [file] [log] [blame]
/*
* Copyright (c) 2019 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.
*/
#ifndef TEST_PC_E2E_TEST_PEER_H_
#define TEST_PC_E2E_TEST_PEER_H_
#include <memory>
#include <string>
#include <vector>
#include "absl/memory/memory.h"
#include "api/test/peerconnection_quality_test_fixture.h"
#include "media/base/media_engine.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/network.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread.h"
#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
#include "test/pc/e2e/peer_connection_quality_test_params.h"
namespace webrtc {
namespace webrtc_pc_e2e {
// Describes a single participant in the call.
class TestPeer final : public PeerConnectionWrapper {
public:
using PeerConnectionWrapper::PeerConnectionWrapper;
using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
struct RemotePeerAudioConfig {
RemotePeerAudioConfig(AudioConfig config)
: sampling_frequency_in_hz(config.sampling_frequency_in_hz),
output_file_name(config.output_dump_file_name) {}
int sampling_frequency_in_hz;
absl::optional<std::string> output_file_name;
};
static absl::optional<RemotePeerAudioConfig> CreateRemoteAudioConfig(
absl::optional<AudioConfig> config);
// Setups all components, that should be provided to WebRTC
// PeerConnectionFactory and PeerConnection creation methods,
// also will setup dependencies, that are required for media analyzers
// injection.
//
// |signaling_thread| will be provided by test fixture implementation.
// |params| - describes current peer paramters, like current peer video
// streams and audio streams
// |audio_outpu_file_name| - the name of output file, where incoming audio
// stream should be written. It should be provided from remote peer
// |params.audio_config.output_file_name|
static std::unique_ptr<TestPeer> CreateTestPeer(
std::unique_ptr<InjectableComponents> components,
std::unique_ptr<Params> params,
std::unique_ptr<MockPeerConnectionObserver> observer,
VideoQualityAnalyzerInjectionHelper* video_analyzer_helper,
rtc::Thread* signaling_thread,
absl::optional<RemotePeerAudioConfig> remote_audio_config,
double bitrate_multiplier,
rtc::TaskQueue* task_queue);
Params* params() const { return params_.get(); }
void DetachAecDump() { audio_processing_->DetachAecDump(); }
// Adds provided |candidates| to the owned peer connection.
bool AddIceCandidates(
std::vector<std::unique_ptr<IceCandidateInterface>> candidates);
private:
TestPeer(rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
rtc::scoped_refptr<PeerConnectionInterface> pc,
std::unique_ptr<MockPeerConnectionObserver> observer,
std::unique_ptr<Params> params,
rtc::scoped_refptr<AudioProcessing> audio_processing);
std::unique_ptr<Params> params_;
rtc::scoped_refptr<AudioProcessing> audio_processing_;
std::vector<std::unique_ptr<IceCandidateInterface>> remote_ice_candidates_;
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_TEST_PEER_H_