// Package webrtc provides the backend to simulate a WebRTC client to send stream
package webrtc

import (
	"bufio"
	"fmt"
	"github.com/pion/rtp"
	"github.com/pion/webrtc/v3"
	"gitlab.crans.org/nounous/ghostream/stream/srt"
	"gitlab.crans.org/nounous/ghostream/stream/telnet"
	"io"
	"log"
	"net"
	"os/exec"
)

var (
	ffmpeg      = make(map[string]*exec.Cmd)
	ffmpegInput = make(map[string]io.WriteCloser)
)

func ingestFrom(inputChannel chan srt.Packet) {
	// FIXME Clean code

	for {
		var err error = nil
		srtPacket := <-inputChannel
		log.Println(len(inputChannel))
		switch srtPacket.PacketType {
		case "register":
			go registerStream(&srtPacket)
			break
		case "sendData":
			if _, ok := ffmpegInput[srtPacket.StreamName]; !ok {
				break
			}
			// FIXME send to stream srtPacket.StreamName
			if _, err := ffmpegInput[srtPacket.StreamName].Write(srtPacket.Data); err != nil {
				log.Printf("Failed to write data to ffmpeg input: %s", err)
			}
			break
		case "close":
			log.Printf("WebRTC CloseConnection %s", srtPacket.StreamName)
			_ = ffmpeg[srtPacket.StreamName].Process.Kill()
			_ = ffmpegInput[srtPacket.StreamName].Close()
			delete(ffmpeg, srtPacket.StreamName)
			delete(ffmpegInput, srtPacket.StreamName)
			break
		default:
			log.Println("Unknown SRT srtPacket type:", srtPacket.PacketType)
			break
		}
		if err != nil {
			log.Printf("Error occured while receiving SRT srtPacket of type %s: %s", srtPacket.PacketType, err)
		}
	}
}

func registerStream(srtPacket *srt.Packet) {
	log.Printf("WebRTC RegisterStream %s", srtPacket.StreamName)

	// Open a UDP Listener for RTP Packets on port 5004
	videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
	if err != nil {
		log.Printf("Faited to open UDP listener %s", err)
		return
	}
	audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5005})
	if err != nil {
		log.Printf("Faited to open UDP listener %s", err)
		return
	}
	// FIXME Close UDP listeners at the end of the stream, not the end of the routine
	/*	defer func() {
		if err = videoListener.Close(); err != nil {
			log.Printf("Faited to close UDP listener %s", err)
		}
		if err = audioListener.Close(); err != nil {
			log.Printf("Faited to close UDP listener %s", err)
		}
	}() */

	ffmpegArgs := []string{"-re", "-i", "pipe:0",
		"-an", "-vcodec", "libvpx", "-crf", "10", "-cpu-used", "5", "-b:v", "6000k", "-maxrate", "8000k", "-bufsize", "12000k", // TODO Change bitrate when changing quality
		"-qmin", "10", "-qmax", "42", "-threads", "4", "-deadline", "1", "-error-resilient", "1",
		"-auto-alt-ref", "1",
		"-f", "rtp", "rtp://127.0.0.1:5004",
		"-vn", "-acodec", "libopus", "-cpu-used", "5", "-deadline", "1", "-qmin", "10", "-qmax", "42", "-error-resilient", "1", "-auto-alt-ref", "1",
		"-f", "rtp", "rtp://127.0.0.1:5005"}

	// Export stream to ascii art
	if telnet.Cfg.Enabled {
		bitrate := fmt.Sprintf("%dk", telnet.Cfg.Width*telnet.Cfg.Height/telnet.Cfg.Delay)
		ffmpegArgs = append(ffmpegArgs,
			"-an", "-vf", fmt.Sprintf("scale=%dx%d", telnet.Cfg.Width, telnet.Cfg.Height),
			"-b:v", bitrate, "-minrate", bitrate, "-maxrate", bitrate, "-bufsize", bitrate, "-q", "42", "-pix_fmt", "gray", "-f", "rawvideo", "pipe:1")
	}

	ffmpeg[srtPacket.StreamName] = exec.Command("ffmpeg", ffmpegArgs...)

	input, err := ffmpeg[srtPacket.StreamName].StdinPipe()
	if err != nil {
		panic(err)
	}
	ffmpegInput[srtPacket.StreamName] = input
	errOutput, err := ffmpeg[srtPacket.StreamName].StderrPipe()
	if err != nil {
		panic(err)
	}

	// Receive raw video output and convert it to ASCII art, then forward it TCP
	if telnet.Cfg.Enabled {
		output, err := ffmpeg[srtPacket.StreamName].StdoutPipe()
		if err != nil {
			panic(err)
		}
		go telnet.StartASCIIArtStream(srtPacket.StreamName, output)
	}

	if err := ffmpeg[srtPacket.StreamName].Start(); err != nil {
		panic(err)
	}

	// Receive video
	go func() {
		inboundRTPPacket := make([]byte, 1500) // UDP MTU
		for {
			n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
			if err != nil {
				log.Printf("Failed to read from UDP: %s", err)
				continue
			}
			packet := &rtp.Packet{}
			if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
				log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
				continue
			}

			if videoTracks[srtPacket.StreamName] == nil {
				videoTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
			}

			// Write RTP srtPacket to all video tracks
			// Adapt payload and SSRC to match destination
			for _, videoTrack := range videoTracks[srtPacket.StreamName] {
				packet.Header.PayloadType = videoTrack.PayloadType()
				packet.Header.SSRC = videoTrack.SSRC()
				if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
					log.Printf("Failed to write to video track: %s", err)
					continue
				}
			}
		}
	}()

	// Receive audio
	go func() {
		inboundRTPPacket := make([]byte, 1500) // UDP MTU
		for {
			n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
			if err != nil {
				log.Printf("Failed to read from UDP: %s", err)
				continue
			}
			packet := &rtp.Packet{}
			if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
				log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
				continue
			}

			if audioTracks[srtPacket.StreamName] == nil {
				audioTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
			}

			// Write RTP srtPacket to all audio tracks
			// Adapt payload and SSRC to match destination
			for _, audioTrack := range audioTracks[srtPacket.StreamName] {
				packet.Header.PayloadType = audioTrack.PayloadType()
				packet.Header.SSRC = audioTrack.SSRC()
				if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
					log.Printf("Failed to write to audio track: %s", err)
					continue
				}
			}
		}
	}()

	go func() {
		scanner := bufio.NewScanner(errOutput)
		for scanner.Scan() {
			log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text())
		}
	}()
}