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/*
* 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 "LibWebRTCDtlsTransportBackend.h"
#include "LibWebRTCPeerConnectionBackend.h"
#include "LibWebRTCRtpSenderTransformBackend.h"
#include "LibWebRTCUtils.h"
#include "RTCPeerConnection.h"
#include "RTCRtpSender.h"
#include "RTCRtpTransformBackend.h"
#include "ScriptExecutionContext.h"
namespace WebCore {
LibWebRTCRtpSenderBackend::LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
: m_peerConnectionBackend(backend)
, m_rtcSender(WTFMove(rtcSender))
{
}
LibWebRTCRtpSenderBackend::LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source)
: m_peerConnectionBackend(backend)
, m_rtcSender(WTFMove(rtcSender))
, m_source(WTFMove(source))
{
startSource();
}
LibWebRTCRtpSenderBackend::~LibWebRTCRtpSenderBackend()
{
stopSource();
}
void LibWebRTCRtpSenderBackend::startSource()
{
// We asynchronously start the sources to guarantee media goes through the transform if a transform is set when creating the track.
callOnMainThread([this, weakThis = WeakPtr { *this }, source = m_source]() mutable {
if (!weakThis)
return;
switchOn(source, [this](Ref<RealtimeOutgoingAudioSource>& source) {
if (auto* currentSource = std::get_if<Ref<RealtimeOutgoingAudioSource>>(&m_source); currentSource && currentSource->ptr() == source.ptr())
source->start();
}, [this](Ref<RealtimeOutgoingVideoSource>& source) {
if (auto* currentSource = std::get_if<Ref<RealtimeOutgoingVideoSource>>(&m_source); currentSource && currentSource->ptr() == source.ptr())
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;
}
bool LibWebRTCRtpSenderBackend::replaceTrack(RTCRtpSender& sender, MediaStreamTrack* track)
{
if (!track) {
stopSource();
return true;
}
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();
});
}
m_peerConnectionBackend->setSenderSourceFromTrack(*this, *track);
return true;
}
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 = std::nullopt;
auto error = m_rtcSender->SetParameters(rtcParameters);
if (!error.ok()) {
promise.reject(toException(error));
return;
}
promise.resolve();
}
std::unique_ptr<RTCDTMFSenderBackend> LibWebRTCRtpSenderBackend::createDTMFBackend()
{
return makeUnique<LibWebRTCDTMFSenderBackend>(m_rtcSender->GetDtmfSender());
}
Ref<RTCRtpTransformBackend> LibWebRTCRtpSenderBackend::rtcRtpTransformBackend()
{
if (!m_transformBackend)
m_transformBackend = LibWebRTCRtpSenderTransformBackend::create(m_rtcSender);
return *m_transformBackend;
}
std::unique_ptr<RTCDtlsTransportBackend> LibWebRTCRtpSenderBackend::dtlsTransportBackend()
{
auto backend = m_rtcSender->dtls_transport();
return backend ? makeUnique<LibWebRTCDtlsTransportBackend>(WTFMove(backend)) : nullptr;
}
void LibWebRTCRtpSenderBackend::setMediaStreamIds(const FixedVector<String>& streamIds)
{
std::vector<std::string> ids;
for (auto& id : streamIds)
ids.push_back(id.utf8().data());
m_rtcSender->SetStreams(ids);
}
RealtimeOutgoingVideoSource* LibWebRTCRtpSenderBackend::videoSource()
{
return WTF::switchOn(m_source,
[](Ref<RealtimeOutgoingVideoSource>& source) { return source.ptr(); },
[](const auto&) -> RealtimeOutgoingVideoSource* { return nullptr; }
);
}
bool LibWebRTCRtpSenderBackend::hasSource() const
{
return WTF::switchOn(m_source,
[](const std::nullptr_t&) { return false; },
[](const auto&) { return true; }
);
}
void LibWebRTCRtpSenderBackend::setSource(Source&& source)
{
stopSource();
m_source = std::exchange(source, nullptr);
startSource();
}
void LibWebRTCRtpSenderBackend::takeSource(LibWebRTCRtpSenderBackend& backend)
{
ASSERT(backend.hasSource());
setSource(WTFMove(backend.m_source));
}
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)