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/*
* Copyright 2011 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.
*/
#ifndef WEBRTC_PC_PEERCONNECTIONFACTORY_H_
#define WEBRTC_PC_PEERCONNECTIONFACTORY_H_
#include <memory>
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/rtc_base/rtccertificategenerator.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread.h"
namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
}
namespace webrtc {
class RtcEventLog;
class PeerConnectionFactory : public PeerConnectionFactoryInterface {
public:
// Use the overloads of CreateVideoSource that take raw VideoCapturer
// pointers from PeerConnectionFactoryInterface.
// TODO(deadbeef): Remove this using statement once those overloads are
// removed.
using PeerConnectionFactoryInterface::CreateVideoSource;
void SetOptions(const Options& options) override;
// Deprecated, use version without constraints.
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) override;
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) override;
bool Initialize();
rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) override;
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) override;
// Deprecated, use version without constraints.
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) override;
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer) override;
// This version supports filtering on width, height and frame rate.
// For the "constraints=null" case, use the version without constraints.
// TODO(hta): Design a version without MediaConstraintsInterface.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer,
const MediaConstraintsInterface* constraints) override;
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& id,
VideoTrackSourceInterface* video_source) override;
rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source) override;
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
// TODO(ivoc) Remove after Chrome is updated.
bool StartRtcEventLog(rtc::PlatformFile file) override { return false; }
// TODO(ivoc) Remove after Chrome is updated.
bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) override {
return false;
}
// TODO(ivoc) Remove after Chrome is updated.
void StopRtcEventLog() override {}
virtual cricket::TransportController* CreateTransportController(
cricket::PortAllocator* port_allocator,
bool redetermine_role_on_ice_restart);
virtual cricket::ChannelManager* channel_manager();
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
virtual rtc::Thread* network_thread();
const Options& options() const { return options_; }
protected:
PeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
rtc::scoped_refptr<AudioMixer> audio_mixer,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
virtual ~PeerConnectionFactory();
private:
std::unique_ptr<RtcEventLog> CreateRtcEventLog_w();
std::unique_ptr<Call> CreateCall_w(RtcEventLog* event_log);
bool wraps_current_thread_;
rtc::Thread* network_thread_;
rtc::Thread* worker_thread_;
rtc::Thread* signaling_thread_;
std::unique_ptr<rtc::Thread> owned_network_thread_;
std::unique_ptr<rtc::Thread> owned_worker_thread_;
Options options_;
// External Audio device used for audio playback.
rtc::scoped_refptr<AudioDeviceModule> default_adm_;
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
// External Video encoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW encoder.
std::unique_ptr<cricket::WebRtcVideoEncoderFactory> video_encoder_factory_;
// External Video decoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW decoder.
std::unique_ptr<cricket::WebRtcVideoDecoderFactory> video_decoder_factory_;
std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
// External audio mixer. This can be NULL. In that case, internal audio mixer
// will be created and used.
rtc::scoped_refptr<AudioMixer> external_audio_mixer_;
std::unique_ptr<cricket::MediaEngineInterface> media_engine_;
std::unique_ptr<webrtc::CallFactoryInterface> call_factory_;
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory_;
};
} // namespace webrtc
#endif // WEBRTC_PC_PEERCONNECTIONFACTORY_H_