blob: befecbf9b2fc90cb330bb4741cda41d2a1c2f1e4 [file] [log] [blame]
/*
* Copyright (C) 2018 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 ENABLE(WEB_RTC)
#include "LibWebRTCMacros.h"
#include "LibWebRTCPeerConnectionBackend.h"
#include "RTCRtpSenderBackend.h"
#include "RealtimeOutgoingAudioSource.h"
#include "RealtimeOutgoingVideoSource.h"
#include <wtf/WeakPtr.h>
ALLOW_UNUSED_PARAMETERS_BEGIN
#include <webrtc/api/rtpsenderinterface.h>
#include <webrtc/rtc_base/scoped_ref_ptr.h>
ALLOW_UNUSED_PARAMETERS_END
namespace WebCore {
class LibWebRTCPeerConnectionBackend;
class LibWebRTCRtpSenderBackend final : public RTCRtpSenderBackend {
WTF_MAKE_FAST_ALLOCATED;
public:
LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
: m_peerConnectionBackend(makeWeakPtr(&backend))
, m_rtcSender(WTFMove(rtcSender))
{
}
using Source = Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>>;
LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source)
: m_peerConnectionBackend(makeWeakPtr(&backend))
, m_rtcSender(WTFMove(rtcSender))
, m_source(WTFMove(source))
{
}
void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); }
RealtimeOutgoingAudioSource* audioSource()
{
return WTF::switchOn(m_source,
[] (Ref<RealtimeOutgoingAudioSource>& source) { return source.ptr(); },
[] (const auto&) -> RealtimeOutgoingAudioSource* { return nullptr; }
);
}
RealtimeOutgoingVideoSource* videoSource()
{
return WTF::switchOn(m_source,
[] (Ref<RealtimeOutgoingVideoSource>& source) { return source.ptr(); },
[] (const auto&) -> RealtimeOutgoingVideoSource* { return nullptr; }
);
}
bool hasSource() const
{
return WTF::switchOn(m_source,
[] (const std::nullptr_t&) { return false; },
[] (const auto&) { return true; }
);
}
void clearSource()
{
ASSERT(hasSource());
m_source = nullptr;
}
void setSource(Source&& source)
{
ASSERT(!hasSource());
m_source = WTFMove(source);
ASSERT(hasSource());
}
void takeSource(LibWebRTCRtpSenderBackend& backend)
{
ASSERT(backend.hasSource());
setSource(WTFMove(backend.m_source));
}
private:
void replaceTrack(ScriptExecutionContext&, RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
RTCRtpSendParameters getParameters() const final;
void setParameters(const RTCRtpSendParameters&, DOMPromiseDeferred<void>&&) final;
std::unique_ptr<RTCDTMFSenderBackend> createDTMFBackend() final;
WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
Source m_source;
mutable Optional<webrtc::RtpParameters> m_currentParameters;
};
} // namespace WebCore
#endif // ENABLE(WEB_RTC)