#ifndef TGVOIP_WEBRTC_MEDIA_MANAGER_H
#define TGVOIP_WEBRTC_MEDIA_MANAGER_H

#include "rtc_base/thread.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "api/transport/field_trial_based_config.h"
#include "pc/rtp_sender.h"

#include <functional>
#include <memory>

namespace webrtc {
class Call;
class RtcEventLogNull;
class TaskQueueFactory;
class VideoBitrateAllocatorFactory;
class VideoTrackSourceInterface;
};

namespace cricket {
class MediaEngineInterface;
class VoiceMediaChannel;
class VideoMediaChannel;
};

#ifdef TGVOIP_NAMESPACE
namespace TGVOIP_NAMESPACE {
#endif

class VideoCapturerInterface;

class MediaManager : public sigslot::has_slots<>, public std::enable_shared_from_this<MediaManager> {
private:
    struct SSRC {
        uint32_t incoming;
        uint32_t outgoing;
        uint32_t fecIncoming;
        uint32_t fecOutgoing;
    };
    
    class NetworkInterfaceImpl : public cricket::MediaChannel::NetworkInterface {
    public:
        NetworkInterfaceImpl(MediaManager *mediaManager, bool isVideo);
        bool SendPacket(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options) override;
        bool SendRtcp(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options) override;
        int SetOption(SocketType type, rtc::Socket::Option opt, int option) override;
        
    private:
        MediaManager *_mediaManager;
        bool _isVideo;
    };
    
    friend class MediaManager::NetworkInterfaceImpl;
    
public:
    MediaManager(
        rtc::Thread *thread,
        bool isOutgoing,
        std::function<void (const rtc::CopyOnWriteBuffer &)> packetEmitted
    );
    ~MediaManager();
    
    void setIsConnected(bool isConnected);
    void receivePacket(const rtc::CopyOnWriteBuffer &packet);
    void notifyPacketSent(const rtc::SentPacket &sentPacket);
    void setIncomingVideoOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
    void setOutgoingVideoOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
    
protected:
    std::function<void (const rtc::CopyOnWriteBuffer &)> _packetEmitted;
    
private:
    rtc::Thread *_thread;
    std::unique_ptr<webrtc::RtcEventLogNull> _eventLog;
    std::unique_ptr<webrtc::TaskQueueFactory> _taskQueueFactory;
    
    SSRC _ssrcAudio;
    SSRC _ssrcVideo;
    
    std::unique_ptr<cricket::MediaEngineInterface> _mediaEngine;
    std::unique_ptr<webrtc::Call> _call;
    webrtc::FieldTrialBasedConfig _fieldTrials;
    webrtc::LocalAudioSinkAdapter _audioSource;
    std::unique_ptr<cricket::VoiceMediaChannel> _audioChannel;
    std::unique_ptr<cricket::VideoMediaChannel> _videoChannel;
    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> _videoBitrateAllocatorFactory;
    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
    std::unique_ptr<VideoCapturerInterface> _videoCapturer;
    std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _currentIncomingVideoSink;
    std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _currentOutgoingVideoSink;
    
    std::unique_ptr<MediaManager::NetworkInterfaceImpl> _audioNetworkInterface;
    std::unique_ptr<MediaManager::NetworkInterfaceImpl> _videoNetworkInterface;
};

#ifdef TGVOIP_NAMESPACE
}
#endif

#endif