| /* |
| * 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) |