| /* |
| * 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() |
| { |
| WTF::switchOn(m_source, [] (Ref<RealtimeOutgoingAudioSource>& source) { |
| source->stop(); |
| }, [] (Ref<RealtimeOutgoingVideoSource>& source) { |
| source->stop(); |
| }, [] (std::nullptr_t&) { |
| }); |
| } |
| |
| template<typename Source> |
| static inline bool updateTrackSource(Source& source, MediaStreamTrack* track) |
| { |
| if (!track) { |
| source.stop(); |
| return true; |
| } |
| return source.setSource(track->privateTrack()); |
| } |
| |
| 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; |
| } |
| |
| auto* currentTrack = sender.track(); |
| |
| ASSERT(!track || !currentTrack || currentTrack->source().type() == track->source().type()); |
| if (currentTrack) { |
| switch (currentTrack->source().type()) { |
| case RealtimeMediaSource::Type::None: |
| ASSERT_NOT_REACHED(); |
| promise.reject(InvalidModificationError); |
| break; |
| case RealtimeMediaSource::Type::Audio: |
| if (!updateTrackSource(*audioSource(), track.get())) { |
| promise.reject(InvalidModificationError); |
| return; |
| } |
| break; |
| case RealtimeMediaSource::Type::Video: |
| if (!updateTrackSource(*videoSource(), track.get())) { |
| promise.reject(InvalidModificationError); |
| return; |
| } |
| break; |
| } |
| } |
| |
| // 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_source = nullptr; |
| 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()); |
| } |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC) |