blob: 3b107076fe8d298ada77deb4610966210a6bf588 [file] [log] [blame]
/*
* Copyright (C) 2017 Apple Inc. All rights reserved.
*
* 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 "LibWebRTCProvider.h"
#include "MediaCapabilitiesDecodingInfo.h"
#include "MediaCapabilitiesEncodingInfo.h"
#include "MediaDecodingConfiguration.h"
#include "MediaEncodingConfiguration.h"
#include "ProcessQualified.h"
#if USE(LIBWEBRTC)
#include "ContentType.h"
#include "LibWebRTCAudioModule.h"
#include "Logging.h"
#include <dlfcn.h>
ALLOW_UNUSED_PARAMETERS_BEGIN
#include <webrtc/api/async_resolver_factory.h>
#include <webrtc/api/audio_codecs/builtin_audio_decoder_factory.h>
#include <webrtc/api/audio_codecs/builtin_audio_encoder_factory.h>
#include <webrtc/api/create_peerconnection_factory.h>
#include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/p2p/base/basic_packet_socket_factory.h>
#include <webrtc/p2p/client/basic_port_allocator.h>
#include <webrtc/pc/peer_connection_factory.h>
#include <webrtc/pc/peer_connection_factory_proxy.h>
#include <webrtc/rtc_base/physical_socket_server.h>
ALLOW_UNUSED_PARAMETERS_END
#include <wtf/Function.h>
#include <wtf/NeverDestroyed.h>
#endif
namespace WebCore {
#if !USE(LIBWEBRTC)
UniqueRef<LibWebRTCProvider> LibWebRTCProvider::create()
{
return makeUniqueRef<LibWebRTCProvider>();
}
bool LibWebRTCProvider::webRTCAvailable()
{
return false;
}
#endif
#if USE(LIBWEBRTC)
LibWebRTCProvider::LibWebRTCProvider()
{
}
#endif
LibWebRTCProvider::~LibWebRTCProvider()
{
}
#if !USE(LIBWEBRTC) || !PLATFORM(COCOA)
void LibWebRTCProvider::registerWebKitVP9Decoder()
{
}
void LibWebRTCProvider::setH264HardwareEncoderAllowed(bool)
{
}
#endif
void LibWebRTCProvider::setActive(bool)
{
}
#if USE(LIBWEBRTC)
static inline rtc::SocketAddress prepareSocketAddress(const rtc::SocketAddress& address, bool disableNonLocalhostConnections)
{
auto result = address;
if (disableNonLocalhostConnections)
result.SetIP("127.0.0.1");
return result;
}
class BasicPacketSocketFactory : public rtc::PacketSocketFactory {
WTF_MAKE_FAST_ALLOCATED;
public:
explicit BasicPacketSocketFactory(rtc::Thread& networkThread)
: m_socketFactory(makeUniqueRefWithoutFastMallocCheck<rtc::BasicPacketSocketFactory>(networkThread.socketserver()))
{
}
void setDisableNonLocalhostConnections(bool disableNonLocalhostConnections) { m_disableNonLocalhostConnections = disableNonLocalhostConnections; }
rtc::AsyncPacketSocket* CreateUdpSocket(const rtc::SocketAddress& address, uint16_t minPort, uint16_t maxPort) final
{
return m_socketFactory->CreateUdpSocket(prepareSocketAddress(address, m_disableNonLocalhostConnections), minPort, maxPort);
}
rtc::AsyncPacketSocket* CreateServerTcpSocket(const rtc::SocketAddress& address, uint16_t minPort, uint16_t maxPort, int options) final
{
return m_socketFactory->CreateServerTcpSocket(prepareSocketAddress(address, m_disableNonLocalhostConnections), minPort, maxPort, options);
}
rtc::AsyncPacketSocket* CreateClientTcpSocket(const rtc::SocketAddress& localAddress, const rtc::SocketAddress& remoteAddress, const rtc::ProxyInfo& info, const std::string& name, const rtc::PacketSocketTcpOptions& options)
{
return m_socketFactory->CreateClientTcpSocket(prepareSocketAddress(localAddress, m_disableNonLocalhostConnections), remoteAddress, info, name, options);
}
rtc::AsyncResolverInterface* CreateAsyncResolver() final { return m_socketFactory->CreateAsyncResolver(); }
private:
bool m_disableNonLocalhostConnections { false };
UniqueRef<rtc::BasicPacketSocketFactory> m_socketFactory;
};
struct PeerConnectionFactoryAndThreads : public rtc::MessageHandler {
std::unique_ptr<rtc::Thread> networkThread;
std::unique_ptr<rtc::Thread> signalingThread;
bool networkThreadWithSocketServer { false };
std::unique_ptr<rtc::NetworkManager> networkManager;
std::unique_ptr<BasicPacketSocketFactory> packetSocketFactory;
std::unique_ptr<rtc::RTCCertificateGenerator> certificateGenerator;
private:
void OnMessage(rtc::Message*);
};
static void doReleaseLogging(rtc::LoggingSeverity severity, const char* message)
{
#if RELEASE_LOG_DISABLED
UNUSED_PARAM(severity);
UNUSED_PARAM(message);
#else
if (severity == rtc::LS_ERROR)
RELEASE_LOG_ERROR(WebRTC, "LibWebRTC error: %" PUBLIC_LOG_STRING, message);
else
RELEASE_LOG(WebRTC, "LibWebRTC message: %" PUBLIC_LOG_STRING, message);
#endif
}
static rtc::LoggingSeverity computeLogLevel(WTFLogLevel level)
{
#if !RELEASE_LOG_DISABLED
switch (level) {
case WTFLogLevel::Always:
case WTFLogLevel::Error:
return rtc::LS_ERROR;
case WTFLogLevel::Warning:
return rtc::LS_WARNING;
case WTFLogLevel::Info:
return rtc::LS_INFO;
case WTFLogLevel::Debug:
return rtc::LS_VERBOSE;
}
ASSERT_NOT_REACHED();
#else
UNUSED_PARAM(level);
#endif
return rtc::LS_NONE;
}
void LibWebRTCProvider::setRTCLogging(WTFLogLevel level)
{
auto rtcLevel = computeLogLevel(level);
rtc::LogMessage::SetLogOutput(rtcLevel, (rtcLevel == rtc::LS_NONE) ? nullptr : doReleaseLogging);
}
static void initializePeerConnectionFactoryAndThreads(PeerConnectionFactoryAndThreads& factoryAndThreads)
{
ASSERT(!factoryAndThreads.networkThread);
factoryAndThreads.networkThread = factoryAndThreads.networkThreadWithSocketServer ? rtc::Thread::CreateWithSocketServer() : rtc::Thread::Create();
factoryAndThreads.networkThread->SetName("WebKitWebRTCNetwork", nullptr);
bool result = factoryAndThreads.networkThread->Start();
ASSERT_UNUSED(result, result);
factoryAndThreads.signalingThread = rtc::Thread::Create();
factoryAndThreads.signalingThread->SetName("WebKitWebRTCSignaling", nullptr);
result = factoryAndThreads.signalingThread->Start();
ASSERT(result);
}
static inline PeerConnectionFactoryAndThreads& staticFactoryAndThreads()
{
static NeverDestroyed<PeerConnectionFactoryAndThreads> factoryAndThreads;
return factoryAndThreads.get();
}
PeerConnectionFactoryAndThreads& LibWebRTCProvider::getStaticFactoryAndThreads(bool useNetworkThreadWithSocketServer)
{
auto& factoryAndThreads = staticFactoryAndThreads();
ASSERT(!factoryAndThreads.networkThread || factoryAndThreads.networkThreadWithSocketServer == useNetworkThreadWithSocketServer);
if (!factoryAndThreads.networkThread) {
factoryAndThreads.networkThreadWithSocketServer = useNetworkThreadWithSocketServer;
initializePeerConnectionFactoryAndThreads(factoryAndThreads);
startedNetworkThread();
}
return factoryAndThreads;
}
struct ThreadMessageData : public rtc::MessageData {
ThreadMessageData(Function<void()>&& callback)
: callback(WTFMove(callback))
{ }
Function<void()> callback;
};
void PeerConnectionFactoryAndThreads::OnMessage(rtc::Message* message)
{
ASSERT(message->message_id == 1);
auto* data = static_cast<ThreadMessageData*>(message->pdata);
data->callback();
delete data;
}
bool LibWebRTCProvider::hasWebRTCThreads()
{
return !!staticFactoryAndThreads().networkThread;
}
void LibWebRTCProvider::callOnWebRTCNetworkThread(Function<void()>&& callback)
{
PeerConnectionFactoryAndThreads& threads = staticFactoryAndThreads();
threads.networkThread->Post(RTC_FROM_HERE, &threads, 1, new ThreadMessageData(WTFMove(callback)));
}
void LibWebRTCProvider::callOnWebRTCSignalingThread(Function<void()>&& callback)
{
PeerConnectionFactoryAndThreads& threads = staticFactoryAndThreads();
threads.signalingThread->Post(RTC_FROM_HERE, &threads, 1, new ThreadMessageData(WTFMove(callback)));
}
rtc::Thread& LibWebRTCProvider::signalingThread()
{
PeerConnectionFactoryAndThreads& threads = staticFactoryAndThreads();
return *threads.signalingThread;
}
void LibWebRTCProvider::setLoggingLevel(WTFLogLevel level)
{
setRTCLogging(level);
}
webrtc::PeerConnectionFactoryInterface* LibWebRTCProvider::factory()
{
if (m_factory)
return m_factory.get();
if (!webRTCAvailable()) {
RELEASE_LOG_ERROR(WebRTC, "LibWebRTC is not available to create a factory");
return nullptr;
}
auto& factoryAndThreads = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer);
m_factory = createPeerConnectionFactory(factoryAndThreads.networkThread.get(), factoryAndThreads.signalingThread.get());
return m_factory;
}
void LibWebRTCProvider::clearFactory()
{
m_audioModule = nullptr;
m_factory = nullptr;
}
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> LibWebRTCProvider::createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread)
{
ASSERT(!m_audioModule);
auto audioModule = rtc::scoped_refptr<LibWebRTCAudioModule>(new rtc::RefCountedObject<LibWebRTCAudioModule>());
m_audioModule = audioModule.get();
return webrtc::CreatePeerConnectionFactory(networkThread, signalingThread, signalingThread, WTFMove(audioModule), webrtc::CreateBuiltinAudioEncoderFactory(), webrtc::CreateBuiltinAudioDecoderFactory(), createEncoderFactory(), createDecoderFactory(), nullptr, nullptr, nullptr);
}
std::unique_ptr<webrtc::VideoDecoderFactory> LibWebRTCProvider::createDecoderFactory()
{
return nullptr;
}
std::unique_ptr<webrtc::VideoEncoderFactory> LibWebRTCProvider::createEncoderFactory()
{
return nullptr;
}
void LibWebRTCProvider::setPeerConnectionFactory(rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>&& factory)
{
auto* thread = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer).signalingThread.get();
m_factory = webrtc::PeerConnectionFactoryProxy::Create(thread, thread, WTFMove(factory));
}
void LibWebRTCProvider::disableEnumeratingAllNetworkInterfaces()
{
m_enableEnumeratingAllNetworkInterfaces = false;
}
void LibWebRTCProvider::enableEnumeratingAllNetworkInterfaces()
{
m_enableEnumeratingAllNetworkInterfaces = true;
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> LibWebRTCProvider::createPeerConnection(ScriptExecutionContextIdentifier, webrtc::PeerConnectionObserver& observer, rtc::PacketSocketFactory*, webrtc::PeerConnectionInterface::RTCConfiguration&& configuration)
{
// Default WK1 implementation.
ASSERT(m_useNetworkThreadWithSocketServer);
auto& factoryAndThreads = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer);
if (!factoryAndThreads.networkManager)
factoryAndThreads.networkManager = makeUniqueWithoutFastMallocCheck<rtc::BasicNetworkManager>();
if (!factoryAndThreads.packetSocketFactory)
factoryAndThreads.packetSocketFactory = makeUnique<BasicPacketSocketFactory>(*factoryAndThreads.networkThread);
factoryAndThreads.packetSocketFactory->setDisableNonLocalhostConnections(m_disableNonLocalhostConnections);
return createPeerConnection(observer, *factoryAndThreads.networkManager, *factoryAndThreads.packetSocketFactory, WTFMove(configuration), nullptr);
}
void LibWebRTCProvider::setEnableWebRTCEncryption(bool enableWebRTCEncryption)
{
auto* factory = this->factory();
if (!factory)
return;
webrtc::PeerConnectionFactoryInterface::Options options;
options.disable_encryption = !enableWebRTCEncryption;
options.ssl_max_version = m_useDTLS10 ? rtc::SSL_PROTOCOL_DTLS_10 : rtc::SSL_PROTOCOL_DTLS_12;
m_factory->SetOptions(options);
}
void LibWebRTCProvider::setUseDTLS10(bool useDTLS10)
{
m_useDTLS10 = useDTLS10;
auto* factory = this->factory();
if (!factory)
return;
webrtc::PeerConnectionFactoryInterface::Options options;
options.ssl_max_version = useDTLS10 ? rtc::SSL_PROTOCOL_DTLS_10 : rtc::SSL_PROTOCOL_DTLS_12;
m_factory->SetOptions(options);
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> LibWebRTCProvider::createPeerConnection(webrtc::PeerConnectionObserver& observer, rtc::NetworkManager& networkManager, rtc::PacketSocketFactory& packetSocketFactory, webrtc::PeerConnectionInterface::RTCConfiguration&& configuration, std::unique_ptr<webrtc::AsyncResolverFactory>&& asyncResolveFactory)
{
auto& factoryAndThreads = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer);
std::unique_ptr<cricket::BasicPortAllocator> portAllocator;
factoryAndThreads.signalingThread->Invoke<void>(RTC_FROM_HERE, [&]() {
auto basicPortAllocator = makeUniqueWithoutFastMallocCheck<cricket::BasicPortAllocator>(&networkManager, &packetSocketFactory);
basicPortAllocator->set_allow_tcp_listen(false);
portAllocator = WTFMove(basicPortAllocator);
});
auto* factory = this->factory();
if (!factory)
return nullptr;
webrtc::PeerConnectionDependencies dependencies { &observer };
dependencies.allocator = WTFMove(portAllocator);
dependencies.async_resolver_factory = WTFMove(asyncResolveFactory);
auto peerConnectionOrError = m_factory->CreatePeerConnectionOrError(configuration, WTFMove(dependencies));
if (!peerConnectionOrError.ok())
return nullptr;
return peerConnectionOrError.MoveValue();
}
void LibWebRTCProvider::prepareCertificateGenerator(Function<void(rtc::RTCCertificateGenerator&)>&& callback)
{
auto& factoryAndThreads = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer);
if (!factoryAndThreads.certificateGenerator)
factoryAndThreads.certificateGenerator = makeUniqueWithoutFastMallocCheck<rtc::RTCCertificateGenerator>(factoryAndThreads.signalingThread.get(), factoryAndThreads.networkThread.get());
auto& generator = *factoryAndThreads.certificateGenerator;
callOnWebRTCSignalingThread([&generator, callback = WTFMove(callback)]() mutable {
callback(generator);
});
}
void LibWebRTCProvider::setH265Support(bool value)
{
m_supportsH265 = value;
m_videoDecodingCapabilities = { };
m_videoEncodingCapabilities = { };
}
void LibWebRTCProvider::setVP9Support(bool supportsVP9Profile0, bool supportsVP9Profile2)
{
m_supportsVP9Profile0 = supportsVP9Profile0;
m_supportsVP9Profile2 = supportsVP9Profile2;
m_videoDecodingCapabilities = { };
m_videoEncodingCapabilities = { };
}
void LibWebRTCProvider::setVP9VTBSupport(bool value)
{
m_supportsVP9VTB = value;
m_videoDecodingCapabilities = { };
m_videoEncodingCapabilities = { };
}
static inline std::optional<cricket::MediaType> typeFromKind(const String& kind)
{
if (kind == "audio"_s)
return cricket::MediaType::MEDIA_TYPE_AUDIO;
if (kind == "video"_s)
return cricket::MediaType::MEDIA_TYPE_VIDEO;
return { };
}
static inline String fromStdString(const std::string& value)
{
return String::fromUTF8(value.data(), value.length());
}
static inline std::optional<uint16_t> toChannels(absl::optional<int> numChannels)
{
if (!numChannels)
return { };
return static_cast<uint32_t>(*numChannels);
}
static inline RTCRtpCapabilities toRTCRtpCapabilities(const webrtc::RtpCapabilities& rtpCapabilities)
{
RTCRtpCapabilities capabilities;
capabilities.codecs.reserveInitialCapacity(rtpCapabilities.codecs.size());
for (auto& codec : rtpCapabilities.codecs) {
StringBuilder sdpFmtpLineBuilder;
bool hasParameter = false;
for (auto& parameter : codec.parameters) {
sdpFmtpLineBuilder.append(hasParameter ? ";" : "", StringView(parameter.first.data(), parameter.first.length()), '=', StringView(parameter.second.data(), parameter.second.length()));
hasParameter = true;
}
String sdpFmtpLine;
if (sdpFmtpLineBuilder.length())
sdpFmtpLine = sdpFmtpLineBuilder.toString();
capabilities.codecs.uncheckedAppend(RTCRtpCodecCapability { fromStdString(codec.mime_type()), static_cast<uint32_t>(codec.clock_rate ? *codec.clock_rate : 0), toChannels(codec.num_channels), WTFMove(sdpFmtpLine) });
}
capabilities.headerExtensions.reserveInitialCapacity(rtpCapabilities.header_extensions.size());
for (auto& header : rtpCapabilities.header_extensions)
capabilities.headerExtensions.uncheckedAppend(RTCRtpCapabilities::HeaderExtensionCapability { fromStdString(header.uri) });
return capabilities;
}
std::optional<RTCRtpCapabilities> LibWebRTCProvider::receiverCapabilities(const String& kind)
{
auto mediaType = typeFromKind(kind);
if (!mediaType)
return { };
switch (*mediaType) {
case cricket::MediaType::MEDIA_TYPE_AUDIO:
return audioDecodingCapabilities();
case cricket::MediaType::MEDIA_TYPE_VIDEO:
return videoDecodingCapabilities();
case cricket::MediaType::MEDIA_TYPE_DATA:
ASSERT_NOT_REACHED();
return { };
case cricket::MediaType::MEDIA_TYPE_UNSUPPORTED:
ASSERT_NOT_REACHED();
return { };
}
}
std::optional<RTCRtpCapabilities>& LibWebRTCProvider::audioDecodingCapabilities()
{
if (!m_audioDecodingCapabilities) {
if (auto* factory = this->factory())
m_audioDecodingCapabilities = toRTCRtpCapabilities(factory->GetRtpReceiverCapabilities(cricket::MediaType::MEDIA_TYPE_AUDIO));
}
return m_audioDecodingCapabilities;
}
std::optional<RTCRtpCapabilities>& LibWebRTCProvider::videoDecodingCapabilities()
{
if (!m_videoDecodingCapabilities) {
if (auto* factory = this->factory())
m_videoDecodingCapabilities = toRTCRtpCapabilities(factory->GetRtpReceiverCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO));
}
return m_videoDecodingCapabilities;
}
std::optional<RTCRtpCapabilities> LibWebRTCProvider::senderCapabilities(const String& kind)
{
auto mediaType = typeFromKind(kind);
if (!mediaType)
return { };
switch (*mediaType) {
case cricket::MediaType::MEDIA_TYPE_AUDIO:
return audioEncodingCapabilities();
case cricket::MediaType::MEDIA_TYPE_VIDEO:
return videoEncodingCapabilities();
case cricket::MediaType::MEDIA_TYPE_DATA:
ASSERT_NOT_REACHED();
return { };
case cricket::MediaType::MEDIA_TYPE_UNSUPPORTED:
ASSERT_NOT_REACHED();
return { };
}
}
std::optional<RTCRtpCapabilities>& LibWebRTCProvider::audioEncodingCapabilities()
{
if (!m_audioEncodingCapabilities) {
if (auto* factory = this->factory())
m_audioEncodingCapabilities = toRTCRtpCapabilities(factory->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_AUDIO));
}
return m_audioEncodingCapabilities;
}
std::optional<RTCRtpCapabilities>& LibWebRTCProvider::videoEncodingCapabilities()
{
if (!m_videoEncodingCapabilities) {
if (auto* factory = this->factory())
m_videoEncodingCapabilities = toRTCRtpCapabilities(factory->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO));
}
return m_videoEncodingCapabilities;
}
std::optional<RTCRtpCodecCapability> LibWebRTCProvider::codecCapability(const ContentType& contentType, const std::optional<RTCRtpCapabilities>& capabilities)
{
if (!capabilities)
return { };
auto containerType = contentType.containerType();
for (auto& codec : capabilities->codecs) {
if (equalIgnoringASCIICase(containerType, codec.mimeType))
return codec;
}
return { };
}
#endif // USE(LIBWEBRTC)
void LibWebRTCProvider::createDecodingConfiguration(MediaDecodingConfiguration&& configuration, DecodingConfigurationCallback&& callback)
{
ASSERT(configuration.type == MediaDecodingType::WebRTC);
#if USE(LIBWEBRTC)
// FIXME: Validate additional parameters, in particular mime type parameters.
MediaCapabilitiesDecodingInfo info { WTFMove(configuration) };
if (info.supportedConfiguration.video) {
ContentType contentType { info.supportedConfiguration.video->contentType };
auto codec = codecCapability(contentType, videoDecodingCapabilities());
if (!codec) {
callback({ });
return;
}
info.supported = true;
#if PLATFORM(COCOA)
auto containerType = contentType.containerType();
if (containerType == "video/vp8")
info.powerEfficient = false;
else if (containerType == "video/vp9")
info.powerEfficient = isSupportingVP9VTB();
else
info.powerEfficient = true;
info.smooth = info.powerEfficient;
#endif
}
if (info.supportedConfiguration.audio) {
ContentType contentType { info.supportedConfiguration.audio->contentType };
auto codec = codecCapability(contentType, audioDecodingCapabilities());
if (!codec) {
callback({ });
return;
}
info.supported = true;
}
callback(WTFMove(info));
#else
UNUSED_PARAM(configuration);
callback({ });
#endif // USE(LIBWEBRTC)
}
void LibWebRTCProvider::createEncodingConfiguration(MediaEncodingConfiguration&& configuration, EncodingConfigurationCallback&& callback)
{
ASSERT(configuration.type == MediaEncodingType::WebRTC);
#if USE(LIBWEBRTC)
// FIXME: Validate additional parameters, in particular mime type parameters.
MediaCapabilitiesEncodingInfo info { WTFMove(configuration) };
if (info.supportedConfiguration.video) {
ContentType contentType { info.supportedConfiguration.video->contentType };
auto codec = codecCapability(contentType, videoEncodingCapabilities());
if (!codec) {
callback({ });
return;
}
info.supported = true;
#if PLATFORM(COCOA)
auto containerType = contentType.containerType();
if (containerType == "video/vp8")
info.powerEfficient = false;
else if (containerType == "video/vp9")
info.powerEfficient = isSupportingVP9VTB();
else
info.powerEfficient = true;
info.smooth = info.powerEfficient;
#endif
}
if (info.supportedConfiguration.audio) {
ContentType contentType { info.supportedConfiguration.audio->contentType };
auto codec = codecCapability(contentType, audioEncodingCapabilities());
if (!codec) {
callback({ });
return;
}
info.supported = true;
}
callback(WTFMove(info));
#else
UNUSED_PARAM(configuration);
callback({ });
#endif // USE(LIBWEBRTC)
}
} // namespace WebCore