diff options
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_factory.cc')
| -rw-r--r-- | third_party/libwebrtc/pc/peer_connection_factory.cc | 359 |
1 files changed, 359 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_factory.cc b/third_party/libwebrtc/pc/peer_connection_factory.cc new file mode 100644 index 0000000000..afebdd79a5 --- /dev/null +++ b/third_party/libwebrtc/pc/peer_connection_factory.cc @@ -0,0 +1,359 @@ +/* + * 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 <type_traits> +#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/logging.h" +#include "rtc_base/numerics/safe_conversions.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->BlockingCall([&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>()), + metronome_(std::move(dependencies->metronome)) {} + +PeerConnectionFactory::PeerConnectionFactory( + PeerConnectionFactoryDependencies dependencies) + : PeerConnectionFactory(ConnectionContext::Create(&dependencies), + &dependencies) {} + +PeerConnectionFactory::~PeerConnectionFactory() { + RTC_DCHECK_RUN_ON(signaling_thread()); + worker_thread()->BlockingCall([this] { + RTC_DCHECK_RUN_ON(worker_thread()); + metronome_ = nullptr; + }); +} + +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; + cricket_codecs = media_engine()->voice().send_codecs(); + auto extensions = + GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice()); + return ToRtpCapabilities(cricket_codecs, extensions); + } + case cricket::MEDIA_TYPE_VIDEO: { + cricket::VideoCodecs cricket_codecs; + cricket_codecs = media_engine()->video().send_codecs(); + auto extensions = + GetDefaultEnabledRtpHeaderExtensions(media_engine()->video()); + return ToRtpCapabilities(cricket_codecs, extensions); + } + 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; + cricket_codecs = media_engine()->voice().recv_codecs(); + auto extensions = + GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice()); + return ToRtpCapabilities(cricket_codecs, extensions); + } + case cricket::MEDIA_TYPE_VIDEO: { + cricket::VideoCodecs cricket_codecs = + media_engine()->video().recv_codecs(context_->use_rtx()); + auto extensions = + GetDefaultEnabledRtpHeaderExtensions(media_engine()->video()); + return ToRtpCapabilities(cricket_codecs, extensions); + } + 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 media_engine()->voice().StartAecDump(FileWrapper(file), + max_size_bytes); +} + +void PeerConnectionFactory::StopAecDump() { + RTC_DCHECK_RUN_ON(worker_thread()); + media_engine()->voice().StopAecDump(); +} + +cricket::MediaEngineInterface* PeerConnectionFactory::media_engine() const { + RTC_DCHECK(context_); + return context_->media_engine(); +} + +RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>> +PeerConnectionFactory::CreatePeerConnectionOrError( + const PeerConnectionInterface::RTCConfiguration& configuration, + PeerConnectionDependencies dependencies) { + RTC_DCHECK_RUN_ON(signaling_thread()); + + // 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) { + const FieldTrialsView* trials = + dependencies.trials ? dependencies.trials.get() : &field_trials(); + dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>( + context_->default_network_manager(), context_->default_socket_factory(), + configuration.turn_customizer, /*relay_port_factory=*/nullptr, trials); + 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()->BlockingCall([this] { return CreateRtcEventLog_w(); }); + + const FieldTrialsView* trials = + dependencies.trials ? dependencies.trials.get() : &field_trials(); + std::unique_ptr<Call> call = + worker_thread()->BlockingCall([this, &event_log, trials, &configuration] { + return CreateCall_w(event_log.get(), *trials, configuration); + }); + + 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, rtc::scoped_refptr<VideoTrackSourceInterface>(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); +} + +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_->Create(encoding_type) + : std::make_unique<RtcEventLogNull>(); +} + +std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w( + RtcEventLog* event_log, + const FieldTrialsView& field_trials, + const PeerConnectionInterface::RTCConfiguration& configuration) { + RTC_DCHECK_RUN_ON(worker_thread()); + + webrtc::Call::Config call_config(event_log, network_thread()); + if (!media_engine() || !context_->call_factory()) { + return nullptr; + } + call_config.audio_state = 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}, + field_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 = &field_trials; + call_config.rtp_transport_controller_send_factory = + transport_controller_send_factory_.get(); + call_config.metronome = metronome_.get(); + call_config.pacer_burst_interval = configuration.pacer_burst_interval; + return std::unique_ptr<Call>( + context_->call_factory()->CreateCall(call_config)); +} + +bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const { + return absl::StartsWith(field_trials().Lookup(key), "Enabled"); +} + +} // namespace webrtc |