Migrate SDP generation to Unified Plan

This commit has breaking changes. This API change means we can no longer support an arbitrary number of receivers. For every track you want to receive you MUST call PeerConnection.AddTransceiver We do now support sending an multiple audio/video feeds. You can see this behavior via gstreamer-receive and gstreamer-send currently. Resolves #54
pion · Apr 4, 2019 · 1202dba · 1202dba
1 parent bc94eaa
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diff --git a/examples/gstreamer-receive/README.md b/examples/gstreamer-receive/README.md
@@ -14,7 +14,7 @@ go get github.com/pions/webrtc/examples/gstreamer-receive
 ```
 
 ### Open gstreamer-receive example page
-[jsfiddle.net](https://jsfiddle.net/pdm7bqfr/) you should see your Webcam, two text-areas and a 'Start Session' button
+[jsfiddle.net](https://jsfiddle.net/8t2g5Lar/) you should see your Webcam, two text-areas and a 'Start Session' button
 
 ### Run gstreamer-receive with your browsers SessionDescription as stdin
 In the jsfiddle the top textarea is your browser, copy that and:

diff --git a/examples/gstreamer-receive/jsfiddle/demo.html b/examples/gstreamer-receive/jsfiddle/demo.html
@@ -4,12 +4,14 @@
 
 Golang base64 Session Description<br />
 <textarea id="remoteSessionDescription"></textarea> <br/>
-<button onclick="window.startSession()"> Start Session </button><br />
+<button onclick="window.startSession()"> Start Session </button>
+<button onclick="window.addDisplayCapture()" id="displayCapture"> Display Capture </button><br />
 
 <br />
 
 Video<br />
-<video id="video1" width="160" height="120" autoplay muted></video> <br />
+<div id="localVideos"></div> <br />
+
 
 Logs<br />
 <div id="logs"></div>
diff --git a/examples/gstreamer-receive/jsfiddle/demo.js b/examples/gstreamer-receive/jsfiddle/demo.js
@@ -7,13 +7,24 @@ let pc = new RTCPeerConnection({
     }
   ]
 })
-var log = msg => {
+let log = msg => {
   document.getElementById('logs').innerHTML += msg + '<br>'
 }
+let displayVideo = video => {
+  var el = document.createElement('video')
+  el.srcObject = video
+  el.autoplay = true
+  el.muted = true
+  el.width = 160
+  el.height = 120
+
+  document.getElementById('localVideos').appendChild(el)
+  return video
+}
 
 navigator.mediaDevices.getUserMedia({ video: true, audio: true })
   .then(stream => {
-    pc.addStream(document.getElementById('video1').srcObject = stream)
+    pc.addStream(displayVideo(stream))
     pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
   }).catch(log)
 
@@ -36,3 +47,11 @@ window.startSession = () => {
     alert(e)
   }
 }
+
+window.addDisplayCapture = () => {
+  navigator.mediaDevices.getDisplayMedia().then(stream => {
+    document.getElementById('displayCapture').disabled = true
+    pc.addStream(displayVideo(stream))
+    pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
+  })
+}
diff --git a/examples/gstreamer-receive/main.go b/examples/gstreamer-receive/main.go
@@ -32,6 +32,15 @@ func gstreamerReceiveMain() {
 		panic(err)
 	}
 
+	// Allow us to receive 1 audio track, and 2 video tracks
+	if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+
 	// Set a handler for when a new remote track starts, this handler creates a gstreamer pipeline
 	// for the given codec
 	peerConnection.OnTrack(func(track *webrtc.Track, receiver *webrtc.RTPReceiver) {

diff --git a/examples/gstreamer-send/README.md b/examples/gstreamer-send/README.md
@@ -14,7 +14,7 @@ go get github.com/pions/webrtc/examples/gstreamer-send
 ```
 
 ### Open gstreamer-send example page
-[jsfiddle.net](https://jsfiddle.net/Laf7ujeo/164/) you should see two text-areas and a 'Start Session' button
+[jsfiddle.net](https://jsfiddle.net/z7ms3u5r/) you should see two text-areas and a 'Start Session' button
 
 ### Run gstreamer-send with your browsers SessionDescription as stdin
 In the jsfiddle the top textarea is your browser, copy that and:

diff --git a/examples/gstreamer-send/jsfiddle/demo.js b/examples/gstreamer-send/jsfiddle/demo.js
@@ -27,7 +27,11 @@ pc.onicecandidate = event => {
   }
 }
 
-pc.createOffer({ offerToReceiveVideo: true, offerToReceiveAudio: true }).then(d => pc.setLocalDescription(d)).catch(log)
+// Offer to receive 1 audio, and 2 video tracks
+pc.addTransceiver('audio', {'direction': 'recvonly'})
+pc.addTransceiver('video', {'direction': 'recvonly'})
+pc.addTransceiver('video', {'direction': 'recvonly'})
+pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
 
 window.startSession = () => {
   let sd = document.getElementById('remoteSessionDescription').value

diff --git a/examples/gstreamer-send/main.go b/examples/gstreamer-send/main.go
@@ -40,21 +40,31 @@ func main() {
 	})
 
 	// Create a audio track
-	opusTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeOpus, rand.Uint32(), "audio", "pion1")
+	audioTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeOpus, rand.Uint32(), "audio", "pion1")
 	if err != nil {
 		panic(err)
 	}
-	_, err = peerConnection.AddTrack(opusTrack)
+	_, err = peerConnection.AddTrack(audioTrack)
 	if err != nil {
 		panic(err)
 	}
 
 	// Create a video track
-	vp8Track, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion2")
+	firstVideoTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion2")
 	if err != nil {
 		panic(err)
 	}
-	_, err = peerConnection.AddTrack(vp8Track)
+	_, err = peerConnection.AddTrack(firstVideoTrack)
+	if err != nil {
+		panic(err)
+	}
+
+	// Create a second video track
+	secondVideoTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion3")
+	if err != nil {
+		panic(err)
+	}
+	_, err = peerConnection.AddTrack(secondVideoTrack)
 	if err != nil {
 		panic(err)
 	}
@@ -85,8 +95,8 @@ func main() {
 	fmt.Println(signal.Encode(answer))
 
 	// Start pushing buffers on these tracks
-	gst.CreatePipeline(webrtc.Opus, []*webrtc.Track{opusTrack}, *audioSrc).Start()
-	gst.CreatePipeline(webrtc.VP8, []*webrtc.Track{vp8Track}, *videoSrc).Start()
+	gst.CreatePipeline(webrtc.Opus, []*webrtc.Track{audioTrack}, *audioSrc).Start()
+	gst.CreatePipeline(webrtc.VP8, []*webrtc.Track{firstVideoTrack, secondVideoTrack}, *videoSrc).Start()
 
 	// Block forever
 	select {}

diff --git a/examples/janus-gateway/streaming/main.go b/examples/janus-gateway/streaming/main.go
@@ -68,6 +68,13 @@ func main() {
 		panic(err)
 	}
 
+	// Allow us to receive 1 audio track, and 1 video track
+	if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+
 	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
 		fmt.Printf("Connection State has changed %s \n", connectionState.String())
 	})

diff --git a/examples/save-to-disk/main.go b/examples/save-to-disk/main.go
@@ -61,6 +61,13 @@ func main() {
 		panic(err)
 	}
 
+	// Allow us to receive 1 audio track, and 1 video track
+	if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+
 	opusFile, err := opuswriter.New("output.opus", 48000, 2)
 	if err != nil {
 		panic(err)

diff --git a/examples/sfu-minimal/README.md b/examples/sfu-minimal/README.md
@@ -10,7 +10,7 @@ go get github.com/pions/webrtc/examples/sfu-minimal
 ```
 
 ### Open sfu-minimal example page
-[jsfiddle.net](https://jsfiddle.net/4g03uqrx/) You should see two buttons 'Publish a Broadcast' and 'Join a Broadcast'
+[jsfiddle.net](https://jsfiddle.net/zhpya3n9/) You should see two buttons 'Publish a Broadcast' and 'Join a Broadcast'
 
 ### Run SFU Minimal
 #### Linux/macOS

diff --git a/examples/sfu-minimal/jsfiddle/demo.js b/examples/sfu-minimal/jsfiddle/demo.js
@@ -27,7 +27,8 @@ window.createSession = isPublisher => {
           .catch(log)
       }).catch(log)
   } else {
-    pc.createOffer({ offerToReceiveVideo: true })
+    pc.addTransceiver('video', {'direction': 'recvonly'})
+    pc.createOffer()
       .then(d => pc.setLocalDescription(d))
       .catch(log)
 

diff --git a/examples/sfu-minimal/main.go b/examples/sfu-minimal/main.go
@@ -2,6 +2,7 @@ package main
 
 import (
 	"fmt"
+	"io"
 	"time"
 
 	"github.com/pions/rtcp"
@@ -46,6 +47,11 @@ func main() {
 		panic(err)
 	}
 
+	// Allow us to receive 1 video track
+	if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+
 	localTrackChan := make(chan *webrtc.Track)
 	// Set a handler for when a new remote track starts, this just distributes all our packets
 	// to connected peers
@@ -75,7 +81,8 @@ func main() {
 				panic(readErr)
 			}
 
-			if _, err = localTrack.Write(rtpBuf[:i]); err != nil {
+			// ErrClosedPipe means we don't have any subscribers, this is ok if no peers have connected yet
+			if _, err = localTrack.Write(rtpBuf[:i]); err != nil && err != io.ErrClosedPipe {
 				panic(err)
 			}
 		}

diff --git a/examples/sfu-ws/room.go b/examples/sfu-ws/room.go
@@ -1,6 +1,7 @@
 package main
 
 import (
+	"io"
 	"net/http"
 	"sync"
 
@@ -69,6 +70,12 @@ func room(w http.ResponseWriter, r *http.Request) {
 		pubReceiver, err = api.NewPeerConnection(peerConnectionConfig)
 		checkError(err)
 
+		_, err = pubReceiver.AddTransceiver(webrtc.RTPCodecTypeAudio)
+		checkError(err)
+
+		_, err = pubReceiver.AddTransceiver(webrtc.RTPCodecTypeVideo)
+		checkError(err)
+
 		pubReceiver.OnTrack(func(remoteTrack *webrtc.Track, receiver *webrtc.RTPReceiver) {
 			if remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeVP8 || remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeVP9 || remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeH264 {
 
@@ -94,7 +101,10 @@ func room(w http.ResponseWriter, r *http.Request) {
 					videoTrackLock.RLock()
 					_, err = videoTrack.Write(rtpBuf[:i])
 					videoTrackLock.RUnlock()
-					checkError(err)
+
+					if err != io.ErrClosedPipe {
+						checkError(err)
+					}
 				}
 
 			} else {
@@ -113,7 +123,9 @@ func room(w http.ResponseWriter, r *http.Request) {
 					audioTrackLock.RLock()
 					_, err = audioTrack.Write(rtpBuf[:i])
 					audioTrackLock.RUnlock()
-					checkError(err)
+					if err != io.ErrClosedPipe {
+						checkError(err)
+					}
 				}
 			}
 		})

diff --git a/examples/sfu-ws/sfu.html b/examples/sfu-ws/sfu.html
@@ -66,7 +66,7 @@
             return alert('Message must not be empty')
         }
         dataChannel.send(message)
-        element.value = '' 
+        element.value = ''
     }
 }
 
@@ -103,7 +103,10 @@
         document.getElementById('msginput').style = 'display: none'
         dataChannel = pc.createDataChannel('data')
         dataChannel.onmessage = e => log(`receive data from '${dataChannel.label}' payload '${e.data}'`)
-        pc.createOffer({ offerToReceiveVideo: true , offerToReceiveAudio: true})
+        pc.addTransceiver('audio', {'direction': 'recvonly'})
+        pc.addTransceiver('video', {'direction': 'recvonly'})
+
+        pc.createOffer()
             .then(d => pc.setLocalDescription(d))
             .catch(log)
 

diff --git a/mediaengine.go b/mediaengine.go
@@ -4,6 +4,7 @@ package webrtc
 
 import (
 	"strconv"
+	"strings"
 
 	"github.com/pions/rtp"
 	"github.com/pions/rtp/codecs"
@@ -164,6 +165,18 @@ func (t RTPCodecType) String() string {
 	}
 }
 
+// NewRTPCodecType creates a RTPCodecType from a string
+func NewRTPCodecType(r string) RTPCodecType {
+	switch {
+	case strings.EqualFold(r, "audio"):
+		return RTPCodecTypeAudio
+	case strings.EqualFold(r, "video"):
+		return RTPCodecTypeVideo
+	default:
+		return RTPCodecType(0)
+	}
+}
+
 // RTPCodec represents a codec supported by the PeerConnection
 type RTPCodec struct {
 	RTPCodecCapability