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/*
* Copyright (C) 2017 Apple Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#pragma once
#if USE(LIBWEBRTC)
#include "LibWebRTCMacros.h"
#include "RTCSignalingState.h"
ALLOW_UNUSED_PARAMETERS_BEGIN
ALLOW_DEPRECATED_DECLARATIONS_BEGIN
#include <webrtc/api/media_stream_interface.h>
#include <webrtc/api/peer_connection_interface.h>
ALLOW_DEPRECATED_DECLARATIONS_END
ALLOW_UNUSED_PARAMETERS_END
#include <wtf/text/WTFString.h>
namespace WebCore {
class LibWebRTCProvider;
class MockRtpSender;
void useMockRTCPeerConnectionFactory(LibWebRTCProvider*, const String&);
void useRealRTCPeerConnectionFactory(LibWebRTCProvider&);
class MockLibWebRTCSessionDescription: public webrtc::SessionDescriptionInterface {
public:
explicit MockLibWebRTCSessionDescription(std::string&& sdp) : m_sdp(WTFMove(sdp)) { }
private:
bool ToString(std::string* out) const final { *out = m_sdp; return true; }
cricket::SessionDescription* description() final { return nullptr; }
const cricket::SessionDescription* description() const final { return nullptr; }
std::string session_id() const final { return ""; }
std::string session_version() const final { return ""; }
std::string type() const final { return ""; }
bool AddCandidate(const webrtc::IceCandidateInterface*) final { return true; }
size_t number_of_mediasections() const final { return 0; }
const webrtc::IceCandidateCollection* candidates(size_t) const final { return nullptr; }
std::string m_sdp;
};
class MockLibWebRTCIceCandidate : public webrtc::IceCandidateInterface {
public:
explicit MockLibWebRTCIceCandidate(const char* sdp, const char* sdpMid)
: m_sdp(sdp)
, m_sdpMid(sdpMid) { }
private:
std::string sdp_mid() const final { return m_sdpMid; }
int sdp_mline_index() const final { return 0; }
const cricket::Candidate& candidate() const final { return m_candidate; }
bool ToString(std::string* out) const final { *out = m_sdp; return true; }
protected:
const char* m_sdp;
const char* m_sdpMid;
cricket::Candidate m_candidate;
};
class MockLibWebRTCAudioTrack : public webrtc::AudioTrackInterface {
public:
explicit MockLibWebRTCAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source)
: m_id(id)
, m_source(source) { }
private:
webrtc::AudioSourceInterface* GetSource() const final { return m_source; }
void AddSink(webrtc::AudioTrackSinkInterface* sink) final {
if (m_source)
m_source->AddSink(sink);
}
void RemoveSink(webrtc::AudioTrackSinkInterface* sink) final {
if (m_source)
m_source->RemoveSink(sink);
}
void RegisterObserver(webrtc::ObserverInterface*) final { }
void UnregisterObserver(webrtc::ObserverInterface*) final { }
std::string kind() const final { return "audio"; }
std::string id() const final { return m_id; }
bool enabled() const final { return m_enabled; }
TrackState state() const final { return kLive; }
bool set_enabled(bool enabled) final { m_enabled = enabled; return true; }
bool m_enabled { true };
std::string m_id;
rtc::scoped_refptr<webrtc::AudioSourceInterface> m_source;
};
class MockLibWebRTCVideoTrack : public webrtc::VideoTrackInterface {
public:
explicit MockLibWebRTCVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source)
: m_id(id)
, m_source(source) { }
private:
webrtc::VideoTrackSourceInterface* GetSource() const final { return m_source; }
void RegisterObserver(webrtc::ObserverInterface*) final { }
void UnregisterObserver(webrtc::ObserverInterface*) final { }
std::string kind() const final { return "video"; }
std::string id() const final { return m_id; }
bool enabled() const final { return m_enabled; }
TrackState state() const final { return kLive; }
bool set_enabled(bool enabled) final { m_enabled = enabled; return true; }
bool m_enabled { true };
std::string m_id;
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> m_source;
};
class MockLibWebRTCDataChannel : public webrtc::DataChannelInterface {
public:
MockLibWebRTCDataChannel(std::string&& label, bool ordered, bool reliable, int id)
: m_label(WTFMove(label))
, m_ordered(ordered)
, m_reliable(reliable)
, m_id(id) { }
private:
void RegisterObserver(webrtc::DataChannelObserver*) final { }
void UnregisterObserver() final { }
std::string label() const final { return m_label; }
bool reliable() const final { return m_reliable; }
bool ordered() const final { return m_ordered; }
int id() const final { return m_id; }
DataState state() const final { return kConnecting; }
uint64_t buffered_amount() const final { return 0; }
void Close() final { }
bool Send(const webrtc::DataBuffer&) final { return true; }
uint32_t messages_sent() const final { return 0; }
uint64_t bytes_sent() const final { return 0; }
uint32_t messages_received() const final { return 0; }
uint64_t bytes_received() const final { return 0; }
std::string m_label;
bool m_ordered { true };
bool m_reliable { false };
int m_id { -1 };
};
class MockDtmfSender : public webrtc::DtmfSenderInterface {
private:
void RegisterObserver(webrtc::DtmfSenderObserverInterface*) final { }
void UnregisterObserver() final { }
bool CanInsertDtmf() final { return false; }
std::string tones() const final { return ""; }
int duration() const final { return 0; }
int inter_tone_gap() const final { return 50; }
};
class MockRtpSender : public webrtc::RtpSenderInterface {
public:
explicit MockRtpSender(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>&& track) : m_track(WTFMove(track)) { }
private:
bool SetTrack(webrtc::MediaStreamTrackInterface* track) final
{
m_track = track;
return true;
}
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const final { return m_track; }
uint32_t ssrc() const { return 0; }
cricket::MediaType media_type() const { return cricket::MEDIA_TYPE_VIDEO; }
std::string id() const { return ""; }
std::vector<std::string> stream_ids() const { return { }; }
webrtc::RtpParameters GetParameters() const final { return { }; }
webrtc::RTCError SetParameters(const webrtc::RtpParameters&) final { return { }; }
rtc::scoped_refptr<webrtc::DtmfSenderInterface> GetDtmfSender() const final
{
if (!m_dtmfSender)
m_dtmfSender = new rtc::RefCountedObject<MockDtmfSender>();
return m_dtmfSender;
}
private:
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> m_track;
mutable rtc::scoped_refptr<webrtc::DtmfSenderInterface> m_dtmfSender;
};
class MockRtpReceiver : public webrtc::RtpReceiverInterface {
private:
cricket::MediaType media_type() const final { return cricket::MEDIA_TYPE_VIDEO; }
std::string id() const { return { }; }
webrtc::RtpParameters GetParameters() const { return { }; }
bool SetParameters(const webrtc::RtpParameters&) { return true; }
void SetObserver(webrtc::RtpReceiverObserverInterface*) { }
void SetJitterBufferMinimumDelay(absl::optional<double>) final { }
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const final
{
if (!m_track)
const_cast<MockRtpReceiver*>(this)->m_track = new rtc::RefCountedObject<MockLibWebRTCVideoTrack>("", nullptr);
return m_track;
}
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> m_track;
};
class MockRtpTransceiver : public webrtc::RtpTransceiverInterface {
public:
MockRtpTransceiver(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& sender, rtc::scoped_refptr<webrtc::RtpReceiverInterface>&& receiver)
: m_sender(WTFMove(sender))
, m_receiver(WTFMove(receiver))
{
}
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender() const final { return m_sender; }
private:
cricket::MediaType media_type() const final { return m_sender->media_type(); }
absl::optional<std::string> mid() const final { return { }; }
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver() const final { return m_receiver; }
bool stopped() const final { return false; }
webrtc::RtpTransceiverDirection direction() const final { return webrtc::RtpTransceiverDirection::kSendRecv; }
void SetDirection(webrtc::RtpTransceiverDirection) final { }
absl::optional<webrtc::RtpTransceiverDirection> current_direction() const final { return { }; }
void StopInternal() final { }
webrtc::RTCError StopStandard() final { return { }; }
private:
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_sender;
rtc::scoped_refptr<webrtc::RtpReceiverInterface> m_receiver;
};
class MockLibWebRTCPeerConnection : public webrtc::PeerConnectionInterface {
public:
~MockLibWebRTCPeerConnection();
protected:
explicit MockLibWebRTCPeerConnection(webrtc::PeerConnectionObserver& observer) : m_observer(observer) { }
private:
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> AddTransceiver(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>) override { return { }; }
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> AddTransceiver(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, const webrtc::RtpTransceiverInit&) override { return { }; }
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> AddTransceiver(cricket::MediaType) override { return { }; }
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> AddTransceiver(cricket::MediaType, const webrtc::RtpTransceiverInit&) override { return { }; }
rtc::scoped_refptr<webrtc::RtpSenderInterface> CreateSender(const std::string&,const std::string&) override { return { }; }
std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> GetSenders() const override { return { }; }
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> GetReceivers() const override { return { }; }
void GetStats(webrtc::RTCStatsCollectorCallback*) override { }
void GetStats(rtc::scoped_refptr<webrtc::RtpSenderInterface>, rtc::scoped_refptr<webrtc::RTCStatsCollectorCallback>) override { }
void GetStats(rtc::scoped_refptr<webrtc::RtpReceiverInterface>, rtc::scoped_refptr<webrtc::RTCStatsCollectorCallback>) override { }
const webrtc::SessionDescriptionInterface* current_local_description() const override { return nullptr; }
const webrtc::SessionDescriptionInterface* current_remote_description() const override { return nullptr; }
const webrtc::SessionDescriptionInterface* pending_local_description() const override { return nullptr; }
const webrtc::SessionDescriptionInterface* pending_remote_description() const override { return nullptr; }
void RestartIce() override { }
webrtc::PeerConnectionInterface::RTCConfiguration GetConfiguration() override { return { }; }
IceConnectionState standardized_ice_connection_state() override { return kIceConnectionNew; }
rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() override { return nullptr; }
rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() override { return nullptr; }
const webrtc::SessionDescriptionInterface* local_description() const override { return nullptr; }
const webrtc::SessionDescriptionInterface* remote_description() const override { return nullptr; }
bool AddIceCandidate(const webrtc::IceCandidateInterface*) override { return true; }
void AddIceCandidate(std::unique_ptr<webrtc::IceCandidateInterface>, std::function<void(webrtc::RTCError)> callback) override { callback({ }); }
SignalingState signaling_state() override;
IceConnectionState ice_connection_state() override { return kIceConnectionNew; }
IceGatheringState ice_gathering_state() override { return kIceGatheringNew; }
void StopRtcEventLog() override { }
void Close() override { }
bool AddStream(webrtc::MediaStreamInterface*) final { return false; }
void RemoveStream(webrtc::MediaStreamInterface*) final { }
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> GetTransceivers() const final;
protected:
void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final { ASSERT_NOT_REACHED(); }
void SetRemoteDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>, rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>) override;
bool RemoveIceCandidates(const std::vector<cricket::Candidate>&) override { return true; }
rtc::scoped_refptr<webrtc::DtlsTransportInterface> LookupDtlsTransportByMid(const std::string&) override { return { }; }
rtc::scoped_refptr<webrtc::SctpTransportInterface> GetSctpTransport() const override { return { }; }
webrtc::PeerConnectionInterface::PeerConnectionState peer_connection_state() override { return PeerConnectionState::kNew; }
bool StartRtcEventLog(std::unique_ptr<webrtc::RtcEventLogOutput>, int64_t) override { return true; }
bool StartRtcEventLog(std::unique_ptr<webrtc::RtcEventLogOutput>) override { return true; }
void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions&) final;
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::DataChannelInterface>> CreateDataChannelOrError(const std::string&, const webrtc::DataChannelInit*) final;
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, const std::vector<std::string>& streams) final;
bool RemoveTrack(webrtc::RtpSenderInterface*) final;
webrtc::RTCError SetBitrate(const webrtc::BitrateSettings&) final { return { }; }
void SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final { ASSERT_NOT_REACHED(); };
void SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>, rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>) override;
bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) override { return false; }
void CreateOffer(webrtc::CreateSessionDescriptionObserver*, const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions&) override;
virtual void gotLocalDescription() { }
webrtc::PeerConnectionObserver& m_observer;
unsigned m_counter { 0 };
Vector<rtc::scoped_refptr<MockRtpTransceiver>> m_transceivers;
bool m_isInitiator { true };
bool m_isReceivingAudio { false };
bool m_isReceivingVideo { false };
std::string m_streamLabel;
RTCSignalingState m_signalingState { RTCSignalingState::Stable };
};
class MockLibWebRTCPeerConnectionFactory : public webrtc::PeerConnectionFactoryInterface {
public:
static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> create(const String& testCase) { return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionFactory>(testCase); }
protected:
explicit MockLibWebRTCPeerConnectionFactory(const String&);
private:
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::PeerConnectionInterface>> CreatePeerConnectionOrError(const webrtc::PeerConnectionInterface::RTCConfiguration&, webrtc::PeerConnectionDependencies) final;
rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateLocalMediaStream(const std::string&) final;
void SetOptions(const Options&) final { }
rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const cricket::AudioOptions&) final { return nullptr; }
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateVideoTrack(const std::string&, webrtc::VideoTrackSourceInterface*) final;
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateAudioTrack(const std::string&, webrtc::AudioSourceInterface*) final;
void StopAecDump() final { }
private:
String m_testCase;
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)