Files
client-sdk-rust/webrtc-sys/include/livekit/peer_connection_factory.h
T
Théo Monnom 33dfcb27b0 fix: datachannel deadlocks & dispose crashes (#87)
- Fix datachannel deadlock on dispose
- Fix audio device being disposed on the wrong thread
- Allow multiple RtcRuntime to be created (Useful for unit tests where we use multiple PeerConnectionFactory)
2023-06-10 18:49:00 +02:00

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2.1 KiB
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/*
* 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.
*/
#pragma once
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
#include "livekit/audio_device.h"
#include "media_stream.h"
#include "peer_connection.h"
#include "rtp_parameters.h"
#include "rust/cxx.h"
#include "webrtc-sys/src/peer_connection.rs.h"
#include "webrtc.h"
namespace livekit {
class PeerConnectionFactory;
} // namespace livekit
#include "webrtc-sys/src/peer_connection_factory.rs.h"
namespace livekit {
webrtc::PeerConnectionInterface::RTCConfiguration to_native_rtc_configuration(
RtcConfiguration config);
class PeerConnectionFactory {
public:
explicit PeerConnectionFactory(std::shared_ptr<RtcRuntime> rtc_runtime);
~PeerConnectionFactory();
std::shared_ptr<PeerConnection> create_peer_connection(
RtcConfiguration config,
std::unique_ptr<NativePeerConnectionObserver> observer) const;
std::shared_ptr<VideoTrack> create_video_track(
rust::String label,
std::shared_ptr<VideoTrackSource> source) const;
std::shared_ptr<AudioTrack> create_audio_track(
rust::String label,
std::shared_ptr<AudioTrackSource> source) const;
RtpCapabilities get_rtp_sender_capabilities(MediaType type) const;
RtpCapabilities get_rtp_receiver_capabilities(MediaType type) const;
private:
std::shared_ptr<RtcRuntime> rtc_runtime_;
rtc::scoped_refptr<AudioDevice> audio_device_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_factory_;
};
std::shared_ptr<PeerConnectionFactory> create_peer_connection_factory();
} // namespace livekit