169 lines
6.4 KiB
C++
169 lines
6.4 KiB
C++
/*
|
|
* Copyright 2023 LiveKit
|
|
*
|
|
* Licensed under the Apache License, Version 2.0 (the “License”);
|
|
* you may not use this file except in compliance with the License.
|
|
* You may obtain a copy of the License at
|
|
*
|
|
* http://www.apache.org/licenses/LICENSE-2.0
|
|
*
|
|
* Unless required by applicable law or agreed to in writing, software
|
|
* distributed under the License is distributed on an “AS IS” BASIS,
|
|
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
|
* See the License for the specific language governing permissions and
|
|
* limitations under the License.
|
|
*/
|
|
|
|
#include "livekit/peer_connection_factory.h"
|
|
|
|
#include <memory>
|
|
#include <utility>
|
|
|
|
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
|
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
|
#include "api/peer_connection_interface.h"
|
|
#include "api/rtc_event_log/rtc_event_log_factory.h"
|
|
#include "api/task_queue/default_task_queue_factory.h"
|
|
#include "api/video_codecs/builtin_video_decoder_factory.h"
|
|
#include "api/video_codecs/builtin_video_encoder_factory.h"
|
|
#include "livekit/audio_device.h"
|
|
#include "livekit/peer_connection.h"
|
|
#include "livekit/rtc_error.h"
|
|
#include "livekit/rtp_parameters.h"
|
|
#include "livekit/video_decoder_factory.h"
|
|
#include "livekit/video_encoder_factory.h"
|
|
#include "livekit/webrtc.h"
|
|
#include "media/engine/webrtc_media_engine.h"
|
|
#include "rtc_base/location.h"
|
|
#include "rtc_base/thread.h"
|
|
#include "webrtc-sys/src/peer_connection.rs.h"
|
|
#include "webrtc-sys/src/peer_connection_factory.rs.h"
|
|
|
|
namespace livekit {
|
|
|
|
webrtc::PeerConnectionInterface::RTCConfiguration to_native_rtc_configuration(
|
|
RtcConfiguration config) {
|
|
webrtc::PeerConnectionInterface::RTCConfiguration rtc_config{};
|
|
|
|
for (auto item : config.ice_servers) {
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.username = item.username.c_str();
|
|
ice_server.password = item.password.c_str();
|
|
|
|
for (auto url : item.urls)
|
|
ice_server.urls.emplace_back(url.c_str());
|
|
|
|
rtc_config.servers.push_back(ice_server);
|
|
}
|
|
|
|
rtc_config.continual_gathering_policy =
|
|
static_cast<webrtc::PeerConnectionInterface::ContinualGatheringPolicy>(
|
|
config.continual_gathering_policy);
|
|
|
|
rtc_config.type =
|
|
static_cast<webrtc::PeerConnectionInterface::IceTransportsType>(
|
|
config.ice_transport_type);
|
|
|
|
return rtc_config;
|
|
}
|
|
|
|
PeerConnectionFactory::PeerConnectionFactory(
|
|
std::shared_ptr<RtcRuntime> rtc_runtime)
|
|
: rtc_runtime_(rtc_runtime) {
|
|
RTC_LOG(LS_INFO) << "PeerConnectionFactory::PeerConnectionFactory()";
|
|
|
|
webrtc::PeerConnectionFactoryDependencies dependencies;
|
|
dependencies.network_thread = rtc_runtime_->network_thread();
|
|
dependencies.worker_thread = rtc_runtime_->worker_thread();
|
|
dependencies.signaling_thread = rtc_runtime_->signaling_thread();
|
|
dependencies.socket_factory = rtc_runtime_->network_thread()->socketserver();
|
|
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
|
|
dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(
|
|
dependencies.task_queue_factory.get());
|
|
dependencies.call_factory = webrtc::CreateCallFactory();
|
|
dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
|
|
|
|
cricket::MediaEngineDependencies media_deps;
|
|
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
|
|
|
|
media_deps.adm = rtc_runtime_->worker_thread()
|
|
->Invoke<rtc::scoped_refptr<livekit::AudioDevice>>(
|
|
RTC_FROM_HERE, [&] {
|
|
return rtc::make_ref_counted<livekit::AudioDevice>(
|
|
media_deps.task_queue_factory);
|
|
});
|
|
|
|
media_deps.video_encoder_factory =
|
|
std::move(std::make_unique<livekit::VideoEncoderFactory>());
|
|
media_deps.video_decoder_factory =
|
|
std::move(std::make_unique<livekit::VideoDecoderFactory>());
|
|
media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
|
|
media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
|
|
media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
|
|
media_deps.trials = dependencies.trials.get();
|
|
|
|
dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
|
|
|
|
peer_factory_ =
|
|
webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
|
|
|
|
if (peer_factory_.get() == nullptr) {
|
|
RTC_LOG_ERR(LS_ERROR) << "Failed to create PeerConnectionFactory";
|
|
return;
|
|
}
|
|
}
|
|
|
|
PeerConnectionFactory::~PeerConnectionFactory() {
|
|
RTC_LOG(LS_INFO) << "PeerConnectionFactory::~PeerConnectionFactory()";
|
|
}
|
|
|
|
std::shared_ptr<PeerConnection> PeerConnectionFactory::create_peer_connection(
|
|
RtcConfiguration config,
|
|
std::unique_ptr<NativePeerConnectionObserver> observer) const {
|
|
observer->rtc_runtime_ = rtc_runtime_; // See peer_connection.h
|
|
webrtc::PeerConnectionDependencies deps{observer.get()};
|
|
auto result = peer_factory_->CreatePeerConnectionOrError(
|
|
to_native_rtc_configuration(config), std::move(deps));
|
|
|
|
if (!result.ok()) {
|
|
throw std::runtime_error(serialize_error(to_error(result.error())));
|
|
}
|
|
|
|
return std::make_shared<PeerConnection>(rtc_runtime_, std::move(observer),
|
|
result.value());
|
|
}
|
|
|
|
std::shared_ptr<VideoTrack> PeerConnectionFactory::create_video_track(
|
|
rust::String label,
|
|
std::shared_ptr<VideoTrackSource> source) const {
|
|
return std::static_pointer_cast<VideoTrack>(
|
|
rtc_runtime_->get_or_create_media_stream_track(
|
|
peer_factory_->CreateVideoTrack(label.c_str(), source->get().get())));
|
|
}
|
|
|
|
std::shared_ptr<AudioTrack> PeerConnectionFactory::create_audio_track(
|
|
rust::String label,
|
|
std::shared_ptr<AudioTrackSource> source) const {
|
|
return std::static_pointer_cast<AudioTrack>(
|
|
rtc_runtime_->get_or_create_media_stream_track(
|
|
peer_factory_->CreateAudioTrack(label.c_str(), source->get().get())));
|
|
}
|
|
|
|
RtpCapabilities PeerConnectionFactory::get_rtp_sender_capabilities(
|
|
MediaType type) const {
|
|
return to_rust_rtp_capabilities(peer_factory_->GetRtpSenderCapabilities(
|
|
static_cast<cricket::MediaType>(type)));
|
|
}
|
|
|
|
RtpCapabilities PeerConnectionFactory::get_rtp_receiver_capabilities(
|
|
MediaType type) const {
|
|
return to_rust_rtp_capabilities(peer_factory_->GetRtpReceiverCapabilities(
|
|
static_cast<cricket::MediaType>(type)));
|
|
}
|
|
|
|
std::shared_ptr<PeerConnectionFactory> create_peer_connection_factory() {
|
|
return std::make_shared<PeerConnectionFactory>(RtcRuntime::create());
|
|
}
|
|
|
|
} // namespace livekit
|