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Sean-DerSean-Der
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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
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‎examples/gstreamer-receive/README.md‎

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -14,7 +14,7 @@ go get github.com/pions/webrtc/examples/gstreamer-receive
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```
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### Open gstreamer-receive example page
17-
[jsfiddle.net](https://jsfiddle.net/pdm7bqfr/) you should see your Webcam, two text-areas and a 'Start Session' button
17+
[jsfiddle.net](https://jsfiddle.net/8t2g5Lar/) you should see your Webcam, two text-areas and a 'Start Session' button
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### Run gstreamer-receive with your browsers SessionDescription as stdin
2020
In the jsfiddle the top textarea is your browser, copy that and:

‎examples/gstreamer-receive/jsfiddle/demo.html‎

Lines changed: 4 additions & 2 deletions
Original file line numberDiff line numberDiff line change
@@ -4,12 +4,14 @@
44

55
Golang base64 Session Description<br />
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<textarea id="remoteSessionDescription"></textarea> <br/>
7-
<button onclick="window.startSession()"> Start Session </button><br />
7+
<button onclick="window.startSession()"> Start Session </button>
8+
<button onclick="window.addDisplayCapture()" id="displayCapture"> Display Capture </button><br />
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<br />
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Video<br />
12-
<video id="video1" width="160" height="120" autoplay muted></video> <br />
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<div id="localVideos"></div> <br />
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1315

1416
Logs<br />
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<div id="logs"></div>

‎examples/gstreamer-receive/jsfiddle/demo.js‎

Lines changed: 21 additions & 2 deletions
Original file line numberDiff line numberDiff line change
@@ -7,13 +7,24 @@ let pc = new RTCPeerConnection({
77
}
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]
99
})
10-
var log = msg => {
10+
let log = msg => {
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document.getElementById('logs').innerHTML += msg + '<br>'
1212
}
13+
let displayVideo = video => {
14+
var el = document.createElement('video')
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el.srcObject = video
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el.autoplay = true
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el.muted = true
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el.width = 160
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el.height = 120
20+
21+
document.getElementById('localVideos').appendChild(el)
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return video
23+
}
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navigator.mediaDevices.getUserMedia({ video: true, audio: true })
1526
.then(stream => {
16-
pc.addStream(document.getElementById('video1').srcObject = stream)
27+
pc.addStream(displayVideo(stream))
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pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
1829
}).catch(log)
1930

@@ -36,3 +47,11 @@ window.startSession = () => {
3647
alert(e)
3748
}
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}
50+
51+
window.addDisplayCapture = () => {
52+
navigator.mediaDevices.getDisplayMedia().then(stream => {
53+
document.getElementById('displayCapture').disabled = true
54+
pc.addStream(displayVideo(stream))
55+
pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
56+
})
57+
}

‎examples/gstreamer-receive/main.go‎

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Original file line numberDiff line numberDiff line change
@@ -32,6 +32,15 @@ func gstreamerReceiveMain() {
3232
panic(err)
3333
}
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35+
// Allow us to receive 1 audio track, and 2 video tracks
36+
if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
37+
panic(err)
38+
} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
39+
panic(err)
40+
} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
41+
panic(err)
42+
}
43+
3544
// Set a handler for when a new remote track starts, this handler creates a gstreamer pipeline
3645
// for the given codec
3746
peerConnection.OnTrack(func(track *webrtc.Track, receiver *webrtc.RTPReceiver) {

‎examples/gstreamer-send/README.md‎

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -14,7 +14,7 @@ go get github.com/pions/webrtc/examples/gstreamer-send
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```
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### Open gstreamer-send example page
17-
[jsfiddle.net](https://jsfiddle.net/Laf7ujeo/164/) you should see two text-areas and a 'Start Session' button
17+
[jsfiddle.net](https://jsfiddle.net/z7ms3u5r/) you should see two text-areas and a 'Start Session' button
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1919
### Run gstreamer-send with your browsers SessionDescription as stdin
2020
In the jsfiddle the top textarea is your browser, copy that and:

‎examples/gstreamer-send/jsfiddle/demo.js‎

Lines changed: 5 additions & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -27,7 +27,11 @@ pc.onicecandidate = event => {
2727
}
2828
}
2929

30-
pc.createOffer({ offerToReceiveVideo: true, offerToReceiveAudio: true }).then(d => pc.setLocalDescription(d)).catch(log)
30+
// Offer to receive 1 audio, and 2 video tracks
31+
pc.addTransceiver('audio', {'direction': 'recvonly'})
32+
pc.addTransceiver('video', {'direction': 'recvonly'})
33+
pc.addTransceiver('video', {'direction': 'recvonly'})
34+
pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
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3236
window.startSession = () => {
3337
let sd = document.getElementById('remoteSessionDescription').value

‎examples/gstreamer-send/main.go‎

Lines changed: 16 additions & 6 deletions
Original file line numberDiff line numberDiff line change
@@ -40,21 +40,31 @@ func main() {
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})
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// Create a audio track
43-
opusTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeOpus, rand.Uint32(), "audio", "pion1")
43+
audioTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeOpus, rand.Uint32(), "audio", "pion1")
4444
if err != nil {
4545
panic(err)
4646
}
47-
_, err = peerConnection.AddTrack(opusTrack)
47+
_, err = peerConnection.AddTrack(audioTrack)
4848
if err != nil {
4949
panic(err)
5050
}
5151

5252
// Create a video track
53-
vp8Track, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion2")
53+
firstVideoTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion2")
5454
if err != nil {
5555
panic(err)
5656
}
57-
_, err = peerConnection.AddTrack(vp8Track)
57+
_, err = peerConnection.AddTrack(firstVideoTrack)
58+
if err != nil {
59+
panic(err)
60+
}
61+
62+
// Create a second video track
63+
secondVideoTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion3")
64+
if err != nil {
65+
panic(err)
66+
}
67+
_, err = peerConnection.AddTrack(secondVideoTrack)
5868
if err != nil {
5969
panic(err)
6070
}
@@ -85,8 +95,8 @@ func main() {
8595
fmt.Println(signal.Encode(answer))
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8797
// Start pushing buffers on these tracks
88-
gst.CreatePipeline(webrtc.Opus, []*webrtc.Track{opusTrack}, *audioSrc).Start()
89-
gst.CreatePipeline(webrtc.VP8, []*webrtc.Track{vp8Track}, *videoSrc).Start()
98+
gst.CreatePipeline(webrtc.Opus, []*webrtc.Track{audioTrack}, *audioSrc).Start()
99+
gst.CreatePipeline(webrtc.VP8, []*webrtc.Track{firstVideoTrack, secondVideoTrack}, *videoSrc).Start()
90100

91101
// Block forever
92102
select {}

‎examples/janus-gateway/streaming/main.go‎

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Original file line numberDiff line numberDiff line change
@@ -68,6 +68,13 @@ func main() {
6868
panic(err)
6969
}
7070

71+
// Allow us to receive 1 audio track, and 1 video track
72+
if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
73+
panic(err)
74+
} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
75+
panic(err)
76+
}
77+
7178
peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
7279
fmt.Printf("Connection State has changed %s \n", connectionState.String())
7380
})

‎examples/save-to-disk/main.go‎

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Original file line numberDiff line numberDiff line change
@@ -61,6 +61,13 @@ func main() {
6161
panic(err)
6262
}
6363

64+
// Allow us to receive 1 audio track, and 1 video track
65+
if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
66+
panic(err)
67+
} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
68+
panic(err)
69+
}
70+
6471
opusFile, err := opuswriter.New("output.opus", 48000, 2)
6572
if err != nil {
6673
panic(err)

‎examples/sfu-minimal/README.md‎

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Original file line numberDiff line numberDiff line change
@@ -10,7 +10,7 @@ go get github.com/pions/webrtc/examples/sfu-minimal
1010
```
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### Open sfu-minimal example page
13-
[jsfiddle.net](https://jsfiddle.net/4g03uqrx/) You should see two buttons 'Publish a Broadcast' and 'Join a Broadcast'
13+
[jsfiddle.net](https://jsfiddle.net/zhpya3n9/) You should see two buttons 'Publish a Broadcast' and 'Join a Broadcast'
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1515
### Run SFU Minimal
1616
#### Linux/macOS

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