snapshot of https://github.com/pion/webrtc.git (main, 2026-09-15; blobs > 131072B omitted)
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29
examples/swap-tracks/README.md
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29
examples/swap-tracks/README.md
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# swap-tracks
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swap-tracks demonstrates how to swap multiple incoming tracks on a single outgoing track.
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## Instructions
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### Download swap-tracks
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```
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go install github.com/pion/webrtc/v4/examples/swap-tracks@latest
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```
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### Open swap-tracks example page
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[jsfiddle.net](https://jsfiddle.net/1rx5on86/) you should see two text-areas and two buttons: `Copy browser SDP to clipboard`, `Start Session`.
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### Run swap-tracks, with your browsers SessionDescription as stdin
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In the jsfiddle the top textarea is your browser's Session Description. Press `Copy browser SDP to clipboard` or copy the base64 string manually.
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We will use this value in the next step.
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#### Linux/macOS
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Run `echo $BROWSER_SDP | swap-tracks`
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#### Windows
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1. Paste the SessionDescription into a file.
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1. Run `swap-tracks < my_file`
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### Input swap-tracks's SessionDescription into your browser
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Copy the text that `swap-tracks` just emitted and copy into second text area
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### Hit 'Start Session' in jsfiddle, enjoy your video!
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Your browser should send streams to Pion, and then a stream will be relayed back, changing every 5 seconds.
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Congrats, you have used Pion WebRTC! Now start building something cool
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8
examples/swap-tracks/jsfiddle/demo.css
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8
examples/swap-tracks/jsfiddle/demo.css
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/*
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SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
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SPDX-License-Identifier: MIT
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*/
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textarea {
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width: 500px;
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min-height: 75px;
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}
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8
examples/swap-tracks/jsfiddle/demo.details
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8
examples/swap-tracks/jsfiddle/demo.details
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---
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# SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
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# SPDX-License-Identifier: MIT
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name: swap-tracks
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description: Example of how to have Pion swap incoming tracks on a single outgoing track
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authors:
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- Chad Retz
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35
examples/swap-tracks/jsfiddle/demo.html
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35
examples/swap-tracks/jsfiddle/demo.html
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<!--
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SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
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SPDX-License-Identifier: MIT
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-->
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Browser base64 Session Description<br />
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<textarea id="localSessionDescription" readonly="true"></textarea> <br />
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<button onclick="window.copySDP()">
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Copy browser SDP to clipboard
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</button>
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<br />
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<br />
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Golang base64 Session Description<br />
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<textarea id="remoteSessionDescription"></textarea> <br/>
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<button onclick="window.startSession()"> Start Session </button><br />
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<br />
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<div style="display: flex">
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<div>
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Browser stream 1<br />
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<canvas id="canvasOne" height="200" width="200"></canvas>
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</div>
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<div>
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Browser stream 2<br />
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<canvas id="canvasTwo" height="200" width="200"></canvas>
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</div>
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<div>
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Browser stream 3<br />
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<canvas id="canvasThree" height="200" width="200"></canvas>
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</div>
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</div>
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Video from server<br />
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<video id="serverVideo" width="200" height="200" autoplay muted></video> <br />
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93
examples/swap-tracks/jsfiddle/demo.js
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examples/swap-tracks/jsfiddle/demo.js
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/* eslint-env browser */
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// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
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// SPDX-License-Identifier: MIT
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// Create peer conn
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const pc = new RTCPeerConnection({
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iceServers: [
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{
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urls: 'stun:stun.l.google.com:19302'
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}
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]
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})
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pc.oniceconnectionstatechange = e => {
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console.debug('connection state change', pc.iceConnectionState)
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}
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pc.onicecandidate = event => {
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if (event.candidate === null) {
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document.getElementById('localSessionDescription').value = btoa(JSON.stringify(pc.localDescription))
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}
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}
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pc.onnegotiationneeded = e =>
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pc.createOffer().then(d => pc.setLocalDescription(d)).catch(console.error)
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pc.ontrack = event => {
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console.log('Got track event', event)
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document.getElementById('serverVideo').srcObject = new MediaStream([event.track])
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}
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const canvases = [
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document.getElementById('canvasOne'),
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document.getElementById('canvasTwo'),
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document.getElementById('canvasThree')
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]
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// Firefox requires getContext to be invoked on an HTML Canvas Element
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// prior to captureStream
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const canvasContexts = canvases.map(c => c.getContext('2d'))
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// Capture canvas streams and add to peer conn
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const streams = canvases.map(c => c.captureStream())
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streams.forEach(stream => stream.getVideoTracks().forEach(track => pc.addTrack(track, stream)))
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// Start circles
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requestAnimationFrame(() => drawCircle(canvasContexts[0], '#006699', 0))
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requestAnimationFrame(() => drawCircle(canvasContexts[1], '#cf635f', 0))
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requestAnimationFrame(() => drawCircle(canvasContexts[2], '#46c240', 0))
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function drawCircle (ctx, color, angle) {
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// Background
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ctx.clearRect(0, 0, 200, 200)
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ctx.fillStyle = '#eeeeee'
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ctx.fillRect(0, 0, 200, 200)
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// Draw and fill in circle
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ctx.beginPath()
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const radius = 25 + 50 * Math.abs(Math.cos(angle))
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ctx.arc(100, 100, radius, 0, Math.PI * 2, false)
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ctx.closePath()
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ctx.fillStyle = color
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ctx.fill()
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// Call again
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requestAnimationFrame(() => drawCircle(ctx, color, angle + (Math.PI / 64)))
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}
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window.startSession = () => {
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const sd = document.getElementById('remoteSessionDescription').value
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if (sd === '') {
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return alert('Session Description must not be empty')
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}
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try {
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pc.setRemoteDescription(JSON.parse(atob(sd)))
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} catch (e) {
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alert(e)
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}
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}
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window.copySDP = () => {
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const browserSDP = document.getElementById('localSessionDescription')
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browserSDP.focus()
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browserSDP.select()
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try {
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const successful = document.execCommand('copy')
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const msg = successful ? 'successful' : 'unsuccessful'
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console.log('Copying SDP was ' + msg)
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} catch (err) {
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console.log('Unable to copy SDP ' + err)
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}
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}
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271
examples/swap-tracks/main.go
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271
examples/swap-tracks/main.go
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@@ -0,0 +1,271 @@
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// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
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// SPDX-License-Identifier: MIT
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//go:build !js
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// swap-tracks demonstrates how to swap multiple incoming tracks on a single outgoing track.
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package main
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import (
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"bufio"
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"context"
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"encoding/base64"
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"encoding/json"
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"errors"
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"fmt"
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"io"
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"os"
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"strings"
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"time"
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"github.com/pion/rtcp"
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"github.com/pion/rtp"
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"github.com/pion/webrtc/v4"
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)
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// nolint: cyclop
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func main() { // nolint:gocognit
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// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
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// Prepare the configuration
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config := webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{
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{
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URLs: []string{"stun:stun.l.google.com:19302"},
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},
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},
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}
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// Create a new RTCPeerConnection
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peerConnection, err := webrtc.NewPeerConnection(config)
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if err != nil {
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panic(err)
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}
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defer func() {
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if cErr := peerConnection.Close(); cErr != nil {
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fmt.Printf("cannot close peerConnection: %v\n", cErr)
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}
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}()
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// Create Track that we send video back to browser on
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outputTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP8,
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}, "video", "pion")
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if err != nil {
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panic(err)
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}
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// Add this newly created track to the PeerConnection
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rtpSender, err := peerConnection.AddTrack(outputTrack)
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if err != nil {
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panic(err)
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}
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// Read incoming RTCP packets
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// Before these packets are returned they are processed by interceptors. For things
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// like NACK this needs to be called.
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
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return
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}
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}
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}()
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// Wait for the offer to be pasted
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offer := webrtc.SessionDescription{}
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decode(readUntilNewline(), &offer)
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// Set the remote SessionDescription
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err = peerConnection.SetRemoteDescription(offer)
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if err != nil {
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panic(err)
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}
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// Which track is currently being handled
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currTrack := 0
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// The total number of tracks
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trackCount := 0
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// The channel of packets with a bit of buffer
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packets := make(chan *rtp.Packet, 60)
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// Set a handler for when a new remote track starts
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peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) { //nolint: revive
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fmt.Printf("Track has started, of type %d: %s \n", track.PayloadType(), track.Codec().MimeType)
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trackNum := trackCount
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trackCount++
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// The last timestamp so that we can change the packet to only be the delta
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var lastTimestamp uint32
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// Whether this track is the one currently sending to the channel (on change
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// of this we send a PLI to have the entire picture updated)
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var isCurrTrack bool
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for {
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// Read RTP packets being sent to Pion
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rtp, _, readErr := track.ReadRTP()
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if readErr != nil {
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panic(readErr)
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}
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// Change the timestamp to only be the delta
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oldTimestamp := rtp.Timestamp
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if lastTimestamp == 0 {
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rtp.Timestamp = 0
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} else {
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rtp.Timestamp -= lastTimestamp
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}
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lastTimestamp = oldTimestamp
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// Check if this is the current track
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if currTrack == trackNum { //nolint:nestif
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// If just switched to this track, send PLI to get picture refresh
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if !isCurrTrack {
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isCurrTrack = true
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if track.Kind() == webrtc.RTPCodecTypeVideo {
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if writeErr := peerConnection.WriteRTCP([]rtcp.Packet{
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&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())},
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}); writeErr != nil {
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fmt.Println(writeErr)
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}
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}
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}
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packets <- rtp
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} else {
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isCurrTrack = false
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}
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}
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})
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ctx, done := context.WithCancel(context.Background())
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// Set the handler for Peer connection state
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// This will notify you when the peer has connected/disconnected
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peerConnection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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fmt.Printf("Peer Connection State has changed: %s\n", state.String())
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if state == webrtc.PeerConnectionStateFailed {
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// Wait until PeerConnection has had no network activity for 30 seconds or another failure.
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// It may be reconnected using an ICE Restart.
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// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
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// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
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done()
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}
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if state == webrtc.PeerConnectionStateClosed {
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// PeerConnection was explicitly closed. This usually happens from a DTLS CloseNotify
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done()
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}
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})
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// Create an answer
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answer, err := peerConnection.CreateAnswer(nil)
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if err != nil {
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panic(err)
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}
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// Create channel that is blocked until ICE Gathering is complete
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gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
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// Sets the LocalDescription, and starts our UDP listeners
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err = peerConnection.SetLocalDescription(answer)
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if err != nil {
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panic(err)
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}
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// Block until ICE Gathering is complete, disabling trickle ICE
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// we do this because we only can exchange one signaling message
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// in a production application you should exchange ICE Candidates via OnICECandidate
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<-gatherComplete
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fmt.Println(encode(peerConnection.LocalDescription()))
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// Asynchronously take all packets in the channel and write them out to our
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// track
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go func() {
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var currTimestamp uint32
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for i := uint16(0); ; i++ {
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packet := <-packets
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// Timestamp on the packet is really a diff, so add it to current
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currTimestamp += packet.Timestamp
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packet.Timestamp = currTimestamp
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// Keep an increasing sequence number
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packet.SequenceNumber = i
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// Write out the packet, ignoring closed pipe if nobody is listening
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if err := outputTrack.WriteRTP(packet); err != nil {
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if errors.Is(err, io.ErrClosedPipe) {
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// The peerConnection has been closed.
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return
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}
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panic(err)
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}
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}
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}()
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// Wait for connection, then rotate the track every 5s
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fmt.Printf("Waiting for connection\n")
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for {
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select {
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case <-ctx.Done():
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return
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default:
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}
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// We haven't gotten any tracks yet
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if trackCount == 0 {
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continue
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}
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fmt.Printf("Waiting 5 seconds then changing...\n")
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time.Sleep(5 * time.Second)
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if currTrack == trackCount-1 {
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currTrack = 0
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} else {
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currTrack++
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}
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fmt.Printf("Switched to track #%v\n", currTrack+1)
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}
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}
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// Read from stdin until we get a newline.
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func readUntilNewline() (in string) {
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var err error
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r := bufio.NewReader(os.Stdin)
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for {
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in, err = r.ReadString('\n')
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if err != nil && !errors.Is(err, io.EOF) {
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panic(err)
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}
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if in = strings.TrimSpace(in); len(in) > 0 {
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break
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}
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}
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fmt.Println("")
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return
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}
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// JSON encode + base64 a SessionDescription.
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func encode(obj *webrtc.SessionDescription) string {
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b, err := json.Marshal(obj)
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if err != nil {
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panic(err)
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}
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return base64.StdEncoding.EncodeToString(b)
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}
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// Decode a base64 and unmarshal JSON into a SessionDescription.
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func decode(in string, obj *webrtc.SessionDescription) {
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b, err := base64.StdEncoding.DecodeString(in)
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if err != nil {
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panic(err)
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}
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if err = json.Unmarshal(b, obj); err != nil {
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panic(err)
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}
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}
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Reference in New Issue
Block a user