| /* |
| * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/peer_connection_factory.h" |
| |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/strings/match.h" |
| #include "api/async_resolver_factory.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/fec_controller.h" |
| #include "api/ice_transport_interface.h" |
| #include "api/network_state_predictor.h" |
| #include "api/packet_socket_factory.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/sequence_checker.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/units/data_rate.h" |
| #include "call/audio_state.h" |
| #include "call/rtp_transport_controller_send_factory.h" |
| #include "media/base/media_engine.h" |
| #include "p2p/base/basic_async_resolver_factory.h" |
| #include "p2p/base/basic_packet_socket_factory.h" |
| #include "p2p/base/default_ice_transport_factory.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/client/basic_port_allocator.h" |
| #include "pc/audio_track.h" |
| #include "pc/local_audio_source.h" |
| #include "pc/media_stream.h" |
| #include "pc/media_stream_proxy.h" |
| #include "pc/media_stream_track_proxy.h" |
| #include "pc/peer_connection.h" |
| #include "pc/peer_connection_factory_proxy.h" |
| #include "pc/peer_connection_proxy.h" |
| #include "pc/rtp_parameters_conversion.h" |
| #include "pc/session_description.h" |
| #include "pc/video_track.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/experiments/field_trial_units.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/system/file_wrapper.h" |
| |
| namespace webrtc { |
| |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> |
| CreateModularPeerConnectionFactory( |
| PeerConnectionFactoryDependencies dependencies) { |
| // The PeerConnectionFactory must be created on the signaling thread. |
| if (dependencies.signaling_thread && |
| !dependencies.signaling_thread->IsCurrent()) { |
| return dependencies.signaling_thread |
| ->Invoke<rtc::scoped_refptr<PeerConnectionFactoryInterface>>( |
| RTC_FROM_HERE, [&dependencies] { |
| return CreateModularPeerConnectionFactory( |
| std::move(dependencies)); |
| }); |
| } |
| |
| auto pc_factory = PeerConnectionFactory::Create(std::move(dependencies)); |
| if (!pc_factory) { |
| return nullptr; |
| } |
| // Verify that the invocation and the initialization ended up agreeing on the |
| // thread. |
| RTC_DCHECK_RUN_ON(pc_factory->signaling_thread()); |
| return PeerConnectionFactoryProxy::Create( |
| pc_factory->signaling_thread(), pc_factory->worker_thread(), pc_factory); |
| } |
| |
| // Static |
| rtc::scoped_refptr<PeerConnectionFactory> PeerConnectionFactory::Create( |
| PeerConnectionFactoryDependencies dependencies) { |
| auto context = ConnectionContext::Create(&dependencies); |
| if (!context) { |
| return nullptr; |
| } |
| return rtc::make_ref_counted<PeerConnectionFactory>(context, &dependencies); |
| } |
| |
| PeerConnectionFactory::PeerConnectionFactory( |
| rtc::scoped_refptr<ConnectionContext> context, |
| PeerConnectionFactoryDependencies* dependencies) |
| : context_(context), |
| task_queue_factory_(std::move(dependencies->task_queue_factory)), |
| event_log_factory_(std::move(dependencies->event_log_factory)), |
| fec_controller_factory_(std::move(dependencies->fec_controller_factory)), |
| network_state_predictor_factory_( |
| std::move(dependencies->network_state_predictor_factory)), |
| injected_network_controller_factory_( |
| std::move(dependencies->network_controller_factory)), |
| neteq_factory_(std::move(dependencies->neteq_factory)), |
| transport_controller_send_factory_( |
| (dependencies->transport_controller_send_factory) |
| ? std::move(dependencies->transport_controller_send_factory) |
| : std::make_unique<RtpTransportControllerSendFactory>()) {} |
| |
| PeerConnectionFactory::PeerConnectionFactory( |
| PeerConnectionFactoryDependencies dependencies) |
| : PeerConnectionFactory(ConnectionContext::Create(&dependencies), |
| &dependencies) {} |
| |
| PeerConnectionFactory::~PeerConnectionFactory() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| } |
| |
| void PeerConnectionFactory::SetOptions(const Options& options) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| options_ = options; |
| } |
| |
| RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities( |
| cricket::MediaType kind) const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| switch (kind) { |
| case cricket::MEDIA_TYPE_AUDIO: { |
| cricket::AudioCodecs cricket_codecs; |
| channel_manager()->GetSupportedAudioSendCodecs(&cricket_codecs); |
| return ToRtpCapabilities( |
| cricket_codecs, |
| channel_manager()->GetDefaultEnabledAudioRtpHeaderExtensions()); |
| } |
| case cricket::MEDIA_TYPE_VIDEO: { |
| cricket::VideoCodecs cricket_codecs; |
| channel_manager()->GetSupportedVideoSendCodecs(&cricket_codecs); |
| return ToRtpCapabilities( |
| cricket_codecs, |
| channel_manager()->GetDefaultEnabledVideoRtpHeaderExtensions()); |
| } |
| case cricket::MEDIA_TYPE_DATA: |
| return RtpCapabilities(); |
| case cricket::MEDIA_TYPE_UNSUPPORTED: |
| return RtpCapabilities(); |
| } |
| RTC_DLOG(LS_ERROR) << "Got unexpected MediaType " << kind; |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities( |
| cricket::MediaType kind) const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| switch (kind) { |
| case cricket::MEDIA_TYPE_AUDIO: { |
| cricket::AudioCodecs cricket_codecs; |
| channel_manager()->GetSupportedAudioReceiveCodecs(&cricket_codecs); |
| return ToRtpCapabilities( |
| cricket_codecs, |
| channel_manager()->GetDefaultEnabledAudioRtpHeaderExtensions()); |
| } |
| case cricket::MEDIA_TYPE_VIDEO: { |
| cricket::VideoCodecs cricket_codecs; |
| channel_manager()->GetSupportedVideoReceiveCodecs(&cricket_codecs); |
| return ToRtpCapabilities( |
| cricket_codecs, |
| channel_manager()->GetDefaultEnabledVideoRtpHeaderExtensions()); |
| } |
| case cricket::MEDIA_TYPE_DATA: |
| return RtpCapabilities(); |
| case cricket::MEDIA_TYPE_UNSUPPORTED: |
| return RtpCapabilities(); |
| } |
| RTC_DLOG(LS_ERROR) << "Got unexpected MediaType " << kind; |
| RTC_CHECK_NOTREACHED(); |
| } |
| |
| rtc::scoped_refptr<AudioSourceInterface> |
| PeerConnectionFactory::CreateAudioSource(const cricket::AudioOptions& options) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| rtc::scoped_refptr<LocalAudioSource> source( |
| LocalAudioSource::Create(&options)); |
| return source; |
| } |
| |
| bool PeerConnectionFactory::StartAecDump(FILE* file, int64_t max_size_bytes) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| return channel_manager()->StartAecDump(FileWrapper(file), max_size_bytes); |
| } |
| |
| void PeerConnectionFactory::StopAecDump() { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| channel_manager()->StopAecDump(); |
| } |
| |
| RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>> |
| PeerConnectionFactory::CreatePeerConnectionOrError( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| PeerConnectionDependencies dependencies) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| RTC_DCHECK(!(dependencies.allocator && dependencies.packet_socket_factory)) |
| << "You can't set both allocator and packet_socket_factory; " |
| "the former is going away (see bugs.webrtc.org/7447"; |
| |
| // Set internal defaults if optional dependencies are not set. |
| if (!dependencies.cert_generator) { |
| dependencies.cert_generator = |
| std::make_unique<rtc::RTCCertificateGenerator>(signaling_thread(), |
| network_thread()); |
| } |
| if (!dependencies.allocator) { |
| rtc::PacketSocketFactory* packet_socket_factory; |
| if (dependencies.packet_socket_factory) |
| packet_socket_factory = dependencies.packet_socket_factory.get(); |
| else |
| packet_socket_factory = context_->default_socket_factory(); |
| |
| dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>( |
| context_->default_network_manager(), packet_socket_factory, |
| configuration.turn_customizer); |
| dependencies.allocator->SetPortRange( |
| configuration.port_allocator_config.min_port, |
| configuration.port_allocator_config.max_port); |
| dependencies.allocator->set_flags( |
| configuration.port_allocator_config.flags); |
| } |
| |
| if (!dependencies.async_resolver_factory) { |
| dependencies.async_resolver_factory = |
| std::make_unique<webrtc::BasicAsyncResolverFactory>(); |
| } |
| |
| if (!dependencies.ice_transport_factory) { |
| dependencies.ice_transport_factory = |
| std::make_unique<DefaultIceTransportFactory>(); |
| } |
| |
| dependencies.allocator->SetNetworkIgnoreMask(options().network_ignore_mask); |
| dependencies.allocator->SetVpnList(configuration.vpn_list); |
| |
| std::unique_ptr<RtcEventLog> event_log = |
| worker_thread()->Invoke<std::unique_ptr<RtcEventLog>>( |
| RTC_FROM_HERE, [this] { return CreateRtcEventLog_w(); }); |
| |
| std::unique_ptr<Call> call = worker_thread()->Invoke<std::unique_ptr<Call>>( |
| RTC_FROM_HERE, |
| [this, &event_log] { return CreateCall_w(event_log.get()); }); |
| |
| auto result = PeerConnection::Create(context_, options_, std::move(event_log), |
| std::move(call), configuration, |
| std::move(dependencies)); |
| if (!result.ok()) { |
| return result.MoveError(); |
| } |
| // We configure the proxy with a pointer to the network thread for methods |
| // that need to be invoked there rather than on the signaling thread. |
| // Internally, the proxy object has a member variable named `worker_thread_` |
| // which will point to the network thread (and not the factory's |
| // worker_thread()). All such methods have thread checks though, so the code |
| // should still be clear (outside of macro expansion). |
| rtc::scoped_refptr<PeerConnectionInterface> result_proxy = |
| PeerConnectionProxy::Create(signaling_thread(), network_thread(), |
| result.MoveValue()); |
| return result_proxy; |
| } |
| |
| rtc::scoped_refptr<MediaStreamInterface> |
| PeerConnectionFactory::CreateLocalMediaStream(const std::string& stream_id) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| return MediaStreamProxy::Create(signaling_thread(), |
| MediaStream::Create(stream_id)); |
| } |
| |
| rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack( |
| const std::string& id, |
| VideoTrackSourceInterface* source) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| rtc::scoped_refptr<VideoTrackInterface> track( |
| VideoTrack::Create(id, source, worker_thread())); |
| return VideoTrackProxy::Create(signaling_thread(), worker_thread(), track); |
| } |
| |
| rtc::scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack( |
| const std::string& id, |
| AudioSourceInterface* source) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| rtc::scoped_refptr<AudioTrackInterface> track( |
| AudioTrack::Create(id, rtc::scoped_refptr<AudioSourceInterface>(source))); |
| return AudioTrackProxy::Create(signaling_thread(), track); |
| } |
| |
| cricket::ChannelManager* PeerConnectionFactory::channel_manager() { |
| return context_->channel_manager(); |
| } |
| |
| std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| |
| auto encoding_type = RtcEventLog::EncodingType::Legacy; |
| if (IsTrialEnabled("WebRTC-RtcEventLogNewFormat")) |
| encoding_type = RtcEventLog::EncodingType::NewFormat; |
| return event_log_factory_ |
| ? event_log_factory_->CreateRtcEventLog(encoding_type) |
| : std::make_unique<RtcEventLogNull>(); |
| } |
| |
| std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w( |
| RtcEventLog* event_log) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| |
| webrtc::Call::Config call_config(event_log, network_thread()); |
| if (!channel_manager()->media_engine() || !context_->call_factory()) { |
| return nullptr; |
| } |
| call_config.audio_state = |
| channel_manager()->media_engine()->voice().GetAudioState(); |
| |
| FieldTrialParameter<DataRate> min_bandwidth("min", |
| DataRate::KilobitsPerSec(30)); |
| FieldTrialParameter<DataRate> start_bandwidth("start", |
| DataRate::KilobitsPerSec(300)); |
| FieldTrialParameter<DataRate> max_bandwidth("max", |
| DataRate::KilobitsPerSec(2000)); |
| ParseFieldTrial({&min_bandwidth, &start_bandwidth, &max_bandwidth}, |
| trials().Lookup("WebRTC-PcFactoryDefaultBitrates")); |
| |
| call_config.bitrate_config.min_bitrate_bps = |
| rtc::saturated_cast<int>(min_bandwidth->bps()); |
| call_config.bitrate_config.start_bitrate_bps = |
| rtc::saturated_cast<int>(start_bandwidth->bps()); |
| call_config.bitrate_config.max_bitrate_bps = |
| rtc::saturated_cast<int>(max_bandwidth->bps()); |
| |
| call_config.fec_controller_factory = fec_controller_factory_.get(); |
| call_config.task_queue_factory = task_queue_factory_.get(); |
| call_config.network_state_predictor_factory = |
| network_state_predictor_factory_.get(); |
| call_config.neteq_factory = neteq_factory_.get(); |
| |
| if (IsTrialEnabled("WebRTC-Bwe-InjectedCongestionController")) { |
| RTC_LOG(LS_INFO) << "Using injected network controller factory"; |
| call_config.network_controller_factory = |
| injected_network_controller_factory_.get(); |
| } else { |
| RTC_LOG(LS_INFO) << "Using default network controller factory"; |
| } |
| |
| call_config.trials = &trials(); |
| call_config.rtp_transport_controller_send_factory = |
| transport_controller_send_factory_.get(); |
| return std::unique_ptr<Call>( |
| context_->call_factory()->CreateCall(call_config)); |
| } |
| |
| bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const { |
| return absl::StartsWith(trials().Lookup(key), "Enabled"); |
| } |
| |
| } // namespace webrtc |