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/*
* Copyright (C) 2017 Apple Inc. All rights reserved.
*
* 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
#include "LibWebRTCMacros.h"
#include "MDNSRegisterError.h"
#include "RTCDataChannelRemoteHandlerConnection.h"
#include "ScriptExecutionContextIdentifier.h"
#include <wtf/CompletionHandler.h>
#include <wtf/Expected.h>
#include <wtf/UniqueRef.h>
#include <wtf/text/WTFString.h>
#if USE(LIBWEBRTC)
#include "RTCRtpCapabilities.h"
ALLOW_UNUSED_PARAMETERS_BEGIN
ALLOW_DEPRECATED_DECLARATIONS_BEGIN
#include <webrtc/api/peer_connection_interface.h>
#include <webrtc/api/video_codecs/video_encoder_factory.h>
#include <webrtc/api/video_codecs/video_decoder_factory.h>
#include <webrtc/api/scoped_refptr.h>
ALLOW_DEPRECATED_DECLARATIONS_END
ALLOW_UNUSED_PARAMETERS_END
namespace rtc {
class NetworkManager;
class PacketSocketFactory;
class Thread;
class RTCCertificateGenerator;
}
namespace webrtc {
class AsyncResolverFactory;
class PeerConnectionFactoryInterface;
}
#endif
namespace WebCore {
class ContentType;
class LibWebRTCAudioModule;
struct MediaCapabilitiesDecodingInfo;
struct MediaCapabilitiesEncodingInfo;
struct MediaDecodingConfiguration;
struct MediaEncodingConfiguration;
class RegistrableDomain;
struct PeerConnectionFactoryAndThreads;
struct RTCRtpCapabilities;
class WEBCORE_EXPORT LibWebRTCProvider {
WTF_MAKE_FAST_ALLOCATED;
public:
static UniqueRef<LibWebRTCProvider> create();
virtual ~LibWebRTCProvider();
static bool webRTCAvailable();
static void registerWebKitVP9Decoder();
static void registerWebKitVP8Decoder();
static void setH264HardwareEncoderAllowed(bool);
static void setRTCLogging(WTFLogLevel);
virtual void setActive(bool);
using IPAddressOrError = Expected<String, MDNSRegisterError>;
using MDNSNameOrError = Expected<String, MDNSRegisterError>;
virtual RefPtr<RTCDataChannelRemoteHandlerConnection> createRTCDataChannelRemoteHandlerConnection() { return nullptr; }
using DecodingConfigurationCallback = Function<void(MediaCapabilitiesDecodingInfo&&)>;
using EncodingConfigurationCallback = Function<void(MediaCapabilitiesEncodingInfo&&)>;
void createDecodingConfiguration(MediaDecodingConfiguration&&, DecodingConfigurationCallback&&);
void createEncodingConfiguration(MediaEncodingConfiguration&&, EncodingConfigurationCallback&&);
#if USE(LIBWEBRTC)
virtual rtc::scoped_refptr<webrtc::PeerConnectionInterface> createPeerConnection(ScriptExecutionContextIdentifier, webrtc::PeerConnectionObserver&, rtc::PacketSocketFactory*, webrtc::PeerConnectionInterface::RTCConfiguration&&);
webrtc::PeerConnectionFactoryInterface* factory();
LibWebRTCAudioModule* audioModule();
// FIXME: Make these methods not static.
static void callOnWebRTCNetworkThread(Function<void()>&&);
static void callOnWebRTCSignalingThread(Function<void()>&&);
static bool hasWebRTCThreads();
static rtc::Thread& signalingThread();
// Used for mock testing
void setPeerConnectionFactory(rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>&&);
void disableEnumeratingAllNetworkInterfaces();
void enableEnumeratingAllNetworkInterfaces();
bool isEnumeratingAllNetworkInterfacesEnabled() const { return m_enableEnumeratingAllNetworkInterfaces; }
void setH265Support(bool);
void setVP9Support(bool supportsVP9Profile0, bool supportsVP9Profile2);
void setVP9VTBSupport(bool);
bool isSupportingH265() const { return m_supportsH265; }
bool isSupportingVP9Profile0() const { return m_supportsVP9Profile0; }
bool isSupportingVP9Profile2() const { return m_supportsVP9Profile2; }
bool isSupportingVP9VTB() const { return m_supportsVP9VTB; }
virtual void disableNonLocalhostConnections() { m_disableNonLocalhostConnections = true; }
bool isSupportingMDNS() const { return m_supportsMDNS; }
// Callback is executed on a background thread.
void prepareCertificateGenerator(Function<void(rtc::RTCCertificateGenerator&)>&&);
std::optional<RTCRtpCapabilities> receiverCapabilities(const String& kind);
std::optional<RTCRtpCapabilities> senderCapabilities(const String& kind);
void clearFactory();
virtual void setLoggingLevel(WTFLogLevel);
void setEnableWebRTCEncryption(bool);
void setUseDTLS10(bool);
class SuspendableSocketFactory : public rtc::PacketSocketFactory {
public:
virtual ~SuspendableSocketFactory() = default;
virtual void suspend() { };
virtual void resume() { };
virtual void disableRelay() { };
};
virtual std::unique_ptr<SuspendableSocketFactory> createSocketFactory(String&& /* userAgent */, bool /* isFirstParty */, RegistrableDomain&&) { return nullptr; }
protected:
LibWebRTCProvider();
rtc::scoped_refptr<webrtc::PeerConnectionInterface> createPeerConnection(webrtc::PeerConnectionObserver&, rtc::NetworkManager&, rtc::PacketSocketFactory&, webrtc::PeerConnectionInterface::RTCConfiguration&&, std::unique_ptr<webrtc::AsyncResolverFactory>&&);
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread);
virtual std::unique_ptr<webrtc::VideoDecoderFactory> createDecoderFactory();
virtual std::unique_ptr<webrtc::VideoEncoderFactory> createEncoderFactory();
virtual void startedNetworkThread() { };
PeerConnectionFactoryAndThreads& getStaticFactoryAndThreads(bool useNetworkThreadWithSocketServer);
std::optional<RTCRtpCapabilities>& audioDecodingCapabilities();
std::optional<RTCRtpCapabilities>& videoDecodingCapabilities();
std::optional<RTCRtpCapabilities>& audioEncodingCapabilities();
std::optional<RTCRtpCapabilities>& videoEncodingCapabilities();
std::optional<RTCRtpCodecCapability> codecCapability(const ContentType&, const std::optional<RTCRtpCapabilities>&);
bool m_enableEnumeratingAllNetworkInterfaces { false };
// FIXME: Remove m_useNetworkThreadWithSocketServer member variable and make it a global.
bool m_useNetworkThreadWithSocketServer { true };
RefPtr<LibWebRTCAudioModule> m_audioModule;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> m_factory;
bool m_disableNonLocalhostConnections { false };
bool m_supportsH265 { false };
bool m_supportsVP9Profile0 { false };
bool m_supportsVP9Profile2 { false };
bool m_supportsVP9VTB { false };
bool m_useDTLS10 { false };
bool m_supportsMDNS { false };
std::optional<RTCRtpCapabilities> m_audioDecodingCapabilities;
std::optional<RTCRtpCapabilities> m_videoDecodingCapabilities;
std::optional<RTCRtpCapabilities> m_audioEncodingCapabilities;
std::optional<RTCRtpCapabilities> m_videoEncodingCapabilities;
#endif
};
#if USE(LIBWEBRTC)
inline LibWebRTCAudioModule* LibWebRTCProvider::audioModule()
{
return m_audioModule.get();
}
#endif
} // namespace WebCore
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