| /* |
| * 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. |
| */ |
| |
| #include "config.h" |
| #include "LibWebRTCRtpSenderBackend.h" |
| |
| #if ENABLE(WEB_RTC) && USE(LIBWEBRTC) |
| |
| #include "JSDOMPromiseDeferred.h" |
| #include "LibWebRTCDTMFSenderBackend.h" |
| #include "LibWebRTCPeerConnectionBackend.h" |
| #include "LibWebRTCUtils.h" |
| #include "RTCPeerConnection.h" |
| #include "RTCRtpSender.h" |
| #include "ScriptExecutionContext.h" |
| |
| namespace WebCore { |
| |
| LibWebRTCRtpSenderBackend::LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source) |
| : m_peerConnectionBackend(makeWeakPtr(&backend)) |
| , m_rtcSender(WTFMove(rtcSender)) |
| , m_source(WTFMove(source)) |
| { |
| startSource(); |
| } |
| |
| LibWebRTCRtpSenderBackend::~LibWebRTCRtpSenderBackend() |
| { |
| stopSource(); |
| } |
| |
| void LibWebRTCRtpSenderBackend::startSource() |
| { |
| switchOn(m_source, [](Ref<RealtimeOutgoingAudioSource>& source) { |
| source->start(); |
| }, [](Ref<RealtimeOutgoingVideoSource>& source) { |
| source->start(); |
| }, [](std::nullptr_t&) { |
| }); |
| } |
| |
| void LibWebRTCRtpSenderBackend::stopSource() |
| { |
| switchOn(m_source, [](Ref<RealtimeOutgoingAudioSource>& source) { |
| source->stop(); |
| }, [](Ref<RealtimeOutgoingVideoSource>& source) { |
| source->stop(); |
| }, [](std::nullptr_t&) { |
| }); |
| m_source = nullptr; |
| } |
| |
| void LibWebRTCRtpSenderBackend::replaceTrack(ScriptExecutionContext& context, RTCRtpSender& sender, RefPtr<MediaStreamTrack>&& track, DOMPromiseDeferred<void>&& promise) |
| { |
| if (!m_peerConnectionBackend) { |
| promise.reject(Exception { InvalidStateError, "No WebRTC backend"_s }); |
| return; |
| } |
| |
| if (!track) |
| stopSource(); |
| else if (sender.track()) { |
| switchOn(m_source, [&](Ref<RealtimeOutgoingAudioSource>& source) { |
| ASSERT(track->source().type() == RealtimeMediaSource::Type::Audio); |
| source->stop(); |
| source->setSource(track->privateTrack()); |
| source->start(); |
| }, [&](Ref<RealtimeOutgoingVideoSource>& source) { |
| ASSERT(track->source().type() == RealtimeMediaSource::Type::Video); |
| source->stop(); |
| source->setSource(track->privateTrack()); |
| source->start(); |
| }, [](std::nullptr_t&) { |
| ASSERT_NOT_REACHED(); |
| }); |
| } |
| |
| // FIXME: Remove this postTask once this whole function is executed as part of the RTCPeerConnection operation queue. |
| context.postTask([protectedSender = makeRef(sender), promise = WTFMove(promise), track = WTFMove(track), this](ScriptExecutionContext&) mutable { |
| if (protectedSender->isStopped()) |
| return; |
| |
| if (!track) { |
| protectedSender->setTrackToNull(); |
| promise.resolve(); |
| return; |
| } |
| |
| bool hasTrack = protectedSender->track(); |
| protectedSender->setTrack(track.releaseNonNull()); |
| |
| if (hasTrack) { |
| promise.resolve(); |
| return; |
| } |
| |
| m_peerConnectionBackend->setSenderSourceFromTrack(*this, *protectedSender->track()); |
| promise.resolve(); |
| }); |
| } |
| |
| RTCRtpSendParameters LibWebRTCRtpSenderBackend::getParameters() const |
| { |
| if (!m_rtcSender) |
| return { }; |
| |
| m_currentParameters = m_rtcSender->GetParameters(); |
| return toRTCRtpSendParameters(*m_currentParameters); |
| } |
| |
| void LibWebRTCRtpSenderBackend::setParameters(const RTCRtpSendParameters& parameters, DOMPromiseDeferred<void>&& promise) |
| { |
| if (!m_rtcSender) { |
| promise.reject(NotSupportedError); |
| return; |
| } |
| |
| if (!m_currentParameters) { |
| promise.reject(Exception { InvalidStateError, "getParameters must be called before setParameters"_s }); |
| return; |
| } |
| |
| auto rtcParameters = WTFMove(*m_currentParameters); |
| updateRTCRtpSendParameters(parameters, rtcParameters); |
| m_currentParameters = WTF::nullopt; |
| |
| auto error = m_rtcSender->SetParameters(rtcParameters); |
| if (!error.ok()) { |
| promise.reject(Exception { InvalidStateError, error.message() }); |
| return; |
| } |
| promise.resolve(); |
| } |
| |
| std::unique_ptr<RTCDTMFSenderBackend> LibWebRTCRtpSenderBackend::createDTMFBackend() |
| { |
| return makeUnique<LibWebRTCDTMFSenderBackend>(m_rtcSender->GetDtmfSender()); |
| } |
| |
| RealtimeOutgoingVideoSource* LibWebRTCRtpSenderBackend::videoSource() |
| { |
| return switchOn(m_source, |
| [](Ref<RealtimeOutgoingVideoSource>& source) { return source.ptr(); }, |
| [](const auto&) -> RealtimeOutgoingVideoSource* { return nullptr; } |
| ); |
| } |
| |
| bool LibWebRTCRtpSenderBackend::hasSource() const |
| { |
| return switchOn(m_source, |
| [](const std::nullptr_t&) { return false; }, |
| [](const auto&) { return true; } |
| ); |
| } |
| |
| void LibWebRTCRtpSenderBackend::setSource(Source&& source) |
| { |
| stopSource(); |
| m_source = WTFMove(source); |
| startSource(); |
| } |
| |
| void LibWebRTCRtpSenderBackend::takeSource(LibWebRTCRtpSenderBackend& backend) |
| { |
| ASSERT(backend.hasSource()); |
| stopSource(); |
| m_source = WTFMove(backend.m_source); |
| backend.m_source = nullptr; |
| } |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC) |