send-audio.go

package main

import (
	"context"
	"encoding/json"
	"errors"
	"fmt"
	"io"
	"net/http"
	"os"
	"signal"
	"time"

	webrtcsignal "signal"

	"github.com/pion/webrtc/v3"
	"github.com/pion/webrtc/v3/pkg/media"
	"github.com/pion/webrtc/v3/pkg/media/oggreader"
)

const (
	oggPageDuration = time.Millisecond * 20
)

func main() {
	resp, err := http.Get("https://cdn.casterlabs.co/api.json")
	if err != nil {
		panic(err)
	}

	body, err := io.ReadAll(resp.Body)
	if err != nil {
		panic(err)
	}

	var api map[string]interface{}
	json.Unmarshal([]byte(body), &api)

	iceServers := api["ice_servers"].([]interface{})
	iceServerList := make([]webrtc.ICEServer, len(iceServers))

	for idx, _server := range iceServers {
		server := _server.(map[string]interface{})

		ice := webrtc.ICEServer{
			URLs: []string{server["urls"].(string)},
		}

		if _, hasCredential := server["credential"]; hasCredential {
			ice.Username = server["username"].(string)
			ice.Credential = server["credential"].(string)
			ice.CredentialType = webrtc.ICECredentialTypePassword
		}

		iceServerList[idx] = ice
	}

	fmt.Fprintf(os.Stderr, "debug:Using ice servers: %s\n", iceServerList)

	// Create a new RTCPeerConnection
	peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{
		ICEServers: iceServerList,
	})

	if err != nil {
		panic(err)
	}

	defer func() {
		peerConnection.Close()
	}()

	iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())

	// Create a audio track
	audioTrack, audioTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, Channels: 2, ClockRate: 48000}, "audio", "pion")
	if audioTrackErr != nil {
		panic(audioTrackErr)
	}

	rtpSender, audioTrackErr := peerConnection.AddTrack(audioTrack)
	if audioTrackErr != nil {
		panic(audioTrackErr)
	}

	// Now that stdin is free, start reading audio.
	go func() {
		// Wait for connection established
		<-iceConnectedCtx.Done()

		// Open on oggfile in non-checksum mode.
		ogg, _, oggErr := oggreader.NewWith(os.Stdin)
		if oggErr != nil {
			panic(oggErr)
		}

		// Keep track of last granule, the difference is the amount of samples in the buffer
		var lastGranule uint64

		// It is important to use a time.Ticker instead of time.Sleep because
		// * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
		// * works around latency issues with Sleep (see https://github.com/golang/go/issues/44343)
		ticker := time.NewTicker(oggPageDuration)
		for ; true; <-ticker.C {
			pageData, pageHeader, oggErr := ogg.ParseNextPage()
			if errors.Is(oggErr, io.EOF) {
				fmt.Fprintln(os.Stderr, "close:")
				os.Exit(0)
			}

			if oggErr != nil {
				panic(oggErr)
			}

			// The amount of samples is the difference between the last and current timestamp
			sampleCount := float64(pageHeader.GranulePosition - lastGranule)
			lastGranule = pageHeader.GranulePosition
			sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond

			if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration}); oggErr != nil {
				panic(oggErr)
			}
		}
	}()

	// Read incoming RTCP packets
	// Before these packets are returned they are processed by interceptors. For things
	// like NACK this needs to be called.
	go func() {
		rtcpBuf := make([]byte, 1500)
		for {
			if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
				return
			}
		}
	}()

	// Set the handler for ICE connection state
	// This will notify you when the peer has connected/disconnected
	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
		fmt.Fprintf(os.Stderr, "state:%s\n", connectionState.String())

		if connectionState == webrtc.ICEConnectionStateConnected {
			fmt.Fprintln(os.Stderr, "ready:")
			iceConnectedCtxCancel()
		}
	})

	// Set the handler for Peer connection state
	// This will notify you when the peer has connected/disconnected
	peerConnection.OnConnectionStateChange(func(s webrtc.PeerConnectionState) {
		fmt.Fprintf(os.Stderr, "peerstate:%s\n", s.String())

		if s == webrtc.PeerConnectionStateFailed {
			// Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
			// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
			// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
			os.Exit(0)
		}
	})

	// Wait for the offer to be pasted
	fmt.Fprintln(os.Stderr, "initialized:")
	offer := webrtc.SessionDescription{}
	signal.Decode(signal.MustReadStdin(), &offer)

	// Set the remote SessionDescription
	if err = peerConnection.SetRemoteDescription(offer); err != nil {
		panic(err)
	}

	// Create answer
	answer, err := peerConnection.CreateAnswer(nil)
	if err != nil {
		panic(err)
	}

	// Create channel that is blocked until ICE Gathering is complete
	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)

	// Sets the LocalDescription, and starts our UDP listeners
	if err = peerConnection.SetLocalDescription(answer); err != nil {
		panic(err)
	}

	// Block until ICE Gathering is complete, disabling trickle ICE
	// we do this because we only can exchange one signaling message
	// in a production application you should exchange ICE Candidates via OnICECandidate
	<-gatherComplete

	// Output the answer in base64
	fmt.Fprintf(os.Stderr, "sdpoffer:%s\n", webrtcsignal.Encode(*peerConnection.LocalDescription()))

	// Block forever
	select {}
}