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/*
* Copyright 2013 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 PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
#define PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
#include <memory>
#include <optional>
#include <string>
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/data_channel_interface.h"
#include "api/field_trials_view.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video/resolution.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_periodic_video_source.h"
#include "pc/test/fake_periodic_video_track_source.h"
#include "pc/test/fake_video_track_renderer.h"
#include "rtc_base/socket_server.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
class PeerConnectionTestWrapper
: public webrtc::PeerConnectionObserver,
public webrtc::CreateSessionDescriptionObserver,
public sigslot::has_slots<> {
public:
static void Connect(PeerConnectionTestWrapper* caller,
PeerConnectionTestWrapper* callee);
PeerConnectionTestWrapper(const std::string& name,
webrtc::SocketServer* socket_server,
webrtc::Thread* network_thread,
webrtc::Thread* worker_thread);
virtual ~PeerConnectionTestWrapper();
bool CreatePc(
const webrtc::PeerConnectionInterface::RTCConfiguration& config,
webrtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
webrtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<webrtc::FieldTrialsView> field_trials = nullptr);
bool CreatePc(
const webrtc::PeerConnectionInterface::RTCConfiguration& config,
webrtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
webrtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
std::unique_ptr<webrtc::FieldTrialsView> field_trials = nullptr);
webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory()
const {
return peer_connection_factory_;
}
webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
webrtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
const std::string& label,
const webrtc::DataChannelInit& init);
std::optional<webrtc::RtpCodecCapability> FindFirstSendCodecWithName(
webrtc::MediaType media_type,
const std::string& name) const;
void WaitForNegotiation();
// Implements PeerConnectionObserver.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override;
void OnAddTrack(
webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
const std::vector<webrtc::scoped_refptr<webrtc::MediaStreamInterface>>&
streams) override;
void OnDataChannel(webrtc::scoped_refptr<webrtc::DataChannelInterface>
data_channel) override;
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
// Implements CreateSessionDescriptionObserver.
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
void OnFailure(webrtc::RTCError) override {}
void CreateOffer(
const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
void CreateAnswer(
const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
void ReceiveOfferSdp(const std::string& sdp);
void ReceiveAnswerSdp(const std::string& sdp);
void AddIceCandidate(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& candidate);
bool WaitForCallEstablished();
bool WaitForConnection();
bool WaitForAudio();
bool WaitForVideo();
void GetAndAddUserMedia(bool audio,
const webrtc::AudioOptions& audio_options,
bool video);
// sigslots
sigslot::signal3<const std::string&, int, const std::string&>
SignalOnIceCandidateReady;
sigslot::signal1<const std::string&> SignalOnSdpReady;
sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
webrtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
bool audio,
const webrtc::AudioOptions& audio_options,
bool video,
webrtc::Resolution resolution = {
.width = webrtc::FakePeriodicVideoSource::kDefaultWidth,
.height = webrtc::FakePeriodicVideoSource::kDefaultHeight});
void StopFakeVideoSources();
private:
void SetLocalDescription(webrtc::SdpType type, const std::string& sdp);
void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp);
bool CheckForConnection();
bool CheckForAudio();
bool CheckForVideo();
std::string name_;
webrtc::SocketServer* const socket_server_;
webrtc::Thread* const network_thread_;
webrtc::Thread* const worker_thread_;
webrtc::SequenceChecker pc_thread_checker_;
webrtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
webrtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
int num_get_user_media_calls_ = 0;
bool pending_negotiation_;
std::vector<webrtc::scoped_refptr<webrtc::FakePeriodicVideoTrackSource>>
fake_video_sources_;
};
#endif // PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
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