Adds media selection to links and webrtc html pages

This commit is contained in:
Alexey Khit
2023-03-10 17:26:46 +03:00
parent a960b9b9ee
commit e0320b8ead
2 changed files with 93 additions and 21 deletions
+51 -5
View File
@@ -73,7 +73,6 @@
<li><a href="api/frame.mp4?src=${src}">frame.mp4</a> snapshot in MP4-format / browsers: all / codecs: H264, H265*</li> <li><a href="api/frame.mp4?src=${src}">frame.mp4</a> snapshot in MP4-format / browsers: all / codecs: H264, H265*</li>
<li><a href="api/stream.m3u8?src=${src}">stream.m3u8</a> HLS/TS / browsers: Safari all, Chrome Android / codecs: H264</li> <li><a href="api/stream.m3u8?src=${src}">stream.m3u8</a> HLS/TS / browsers: Safari all, Chrome Android / codecs: H264</li>
<li><a href="api/stream.m3u8?src=${src}&mp4">stream.m3u8</a> HLS/fMP4 / browsers: Safari all, Chrome Android / codecs: H264, H265*, AAC</li> <li><a href="api/stream.m3u8?src=${src}&mp4">stream.m3u8</a> HLS/fMP4 / browsers: Safari all, Chrome Android / codecs: H264, H265*, AAC</li>
<li><a href="webrtc.html?src=${src}">webrtc.html</a> with two-way audio for supported cameras / browsers: all / codecs: H264, PCMU, PCMA, OPUS</li>
<h2>MJPEG source</h2> <h2>MJPEG source</h2>
<li><a href="stream.html?src=${src}&mode=mjpeg">stream.html</a> with MJPEG mode / browsers: all / codecs: MJPEG, JPEG</li> <li><a href="stream.html?src=${src}&mode=mjpeg">stream.html</a> with MJPEG mode / browsers: all / codecs: MJPEG, JPEG</li>
@@ -85,6 +84,7 @@
<div> <div>
<h2>Play audio</h2> <h2>Play audio</h2>
<pre>example: ffmpeg:https://example.com/song.mp3#audio=pcma#input=file</pre>
<input id="source" type="text" placeholder="source"> <input id="source" type="text" placeholder="source">
<a id="send" href="#">send</a> / cameras with two way audio support <a id="send" href="#">send</a> / cameras with two way audio support
</div> </div>
@@ -98,13 +98,50 @@
}) })
</script> </script>
<div> <div id="webrtc">
<h2>Share stream</h2> <h2>WebRTC Magic</h2>
<a id="shareadd" href="#">share</a> <li>video+audio+microphone = two way audio camera</li>
<li>camera+microphone+audio = two way audio browser</li>
<li>display = broadcaster software</li>
<br>
<label><input type="radio" name="video" value="video" checked>remote video</label>
<label><input type="radio" name="video" value="camera">local camera</label>
<label><input type="radio" name="video" value="display">local display</label>
<label><input type="radio" name="video" value="">none</label>
<br>
<label><input type="radio" name="audio1" value="audio" checked>remote audio</label>
<label><input type="radio" name="audio1" value="">none</label>
<br>
<label><input type="radio" name="audio2" value="microphone" checked>local microphone</label>
<label><input type="radio" name="audio2" value="speaker">local speaker</label>
<label><input type="radio" name="audio2" value="">none</label>
<br>
<br>
<li><a id="local" href="webrtc.html?src=">webrtc.html</a> local WebRTC viewer</li>
<li>
<a id="shareadd" href="#">share link</a>
<a id="shareget" href="#">copy link</a> <a id="shareget" href="#">copy link</a>
<a id="sharedel" href="#">delete</a> <a id="sharedel" href="#">delete</a>
external WebRTC viewer
</li>
</div> </div>
<script> <script>
function webrtcLinksUpdate() {
let media = document.querySelectorAll('#webrtc input')
media = Array.from(media).filter(i => i.value && i.checked).map(i => i.value).join('+')
const direction = media.indexOf('video') >= 0 || media === 'audio' ? 'src' : 'dst'
document.getElementById('local').href = `webrtc.html?${direction}=${src}&media=${media}`
const share = document.getElementById('shareget')
share.href = `https://alexxit.github.io/go2rtc/#${share.dataset.auth}&media=${media}`
}
function share(method) { function share(method) {
const url = new URL('api/webtorrent', location.href) const url = new URL('api/webtorrent', location.href)
url.searchParams.set('src', src) url.searchParams.set('src', src)
@@ -112,10 +149,13 @@
} }
function onshareadd(r) { function onshareadd(r) {
document.getElementById('shareget').dataset['auth'] = `share=${r.share}&pwd=${r.pwd}`
document.getElementById('shareadd').style.display = 'none' document.getElementById('shareadd').style.display = 'none'
document.getElementById('shareget').style.display = '' document.getElementById('shareget').style.display = ''
document.getElementById('sharedel').style.display = '' document.getElementById('sharedel').style.display = ''
document.getElementById('shareget').href = `https://alexxit.github.io/go2rtc/#share=${r.share}&pwd=${r.pwd}`
webrtcLinksUpdate()
} }
function onsharedel() { function onsharedel() {
@@ -139,10 +179,16 @@
share('DELETE').then(r => onsharedel()) share('DELETE').then(r => onsharedel())
}) })
document.getElementById('webrtc').addEventListener('click', ev => {
if (ev.target.tagName === 'INPUT') webrtcLinksUpdate()
})
share('GET').then(r => { share('GET').then(r => {
if (r.ok) r.json().then(r => onshareadd(r)) if (r.ok) r.json().then(r => onshareadd(r))
else onsharedel() else onsharedel()
}) })
webrtcLinksUpdate()
</script> </script>
</body> </body>
+40 -14
View File
@@ -19,36 +19,61 @@
<body> <body>
<video id="video" autoplay controls playsinline muted></video> <video id="video" autoplay controls playsinline muted></video>
<script> <script>
function PeerConnection(userMedia) { async function PeerConnection(media) {
const pc = new RTCPeerConnection({ const pc = new RTCPeerConnection({
iceServers: [{urls: 'stun:stun.l.google.com:19302'}] iceServers: [{urls: 'stun:stun.l.google.com:19302'}]
}) })
document.getElementById('video').srcObject = new MediaStream([ const localTracks = []
pc.addTransceiver('video', {direction: 'recvonly'}).receiver.track,
pc.addTransceiver('audio', {direction: 'recvonly'}).receiver.track
])
if (userMedia) { if (/video|audio/.test(media)) {
userMedia.getTracks().forEach(track => { const tracks = ['video', 'audio']
.filter(kind => media.indexOf(kind) >= 0)
.map(kind => pc.addTransceiver(kind, {direction: 'recvonly'}).receiver.track)
localTracks.push(...tracks)
}
if (/camera|microphone/.test(media)) {
const tracks = await getMediaTracks('user', {
video: media.indexOf('camera') >= 0,
audio: media.indexOf('microphone') >= 0,
})
tracks.forEach(track => {
pc.addTransceiver(track, {direction: 'sendonly'}) pc.addTransceiver(track, {direction: 'sendonly'})
if (track.kind === 'video') localTracks.push(track)
}) })
} }
if (media.indexOf('display') >= 0) {
const tracks = await getMediaTracks('display', {
video: true,
audio: media.indexOf('speaker') >= 0,
})
tracks.forEach(track => {
pc.addTransceiver(track, {direction: 'sendonly'})
if (track.kind === 'video') localTracks.push(track)
})
}
document.getElementById('video').srcObject = new MediaStream(localTracks)
return pc return pc
} }
async function userMedia() { async function getMediaTracks(media, constraints) {
try { try {
return await navigator.mediaDevices.getUserMedia({audio: true}) const stream = media === 'user'
? await navigator.mediaDevices.getUserMedia(constraints)
: await navigator.mediaDevices.getDisplayMedia(constraints)
return stream.getTracks()
} catch (e) { } catch (e) {
return null console.warn(e)
return []
} }
} }
async function connect() { async function connect(media) {
const pc = PeerConnection(await userMedia()) const pc = await PeerConnection(media)
const url = new URL('api/ws' + location.search, location.href) const url = new URL('api/ws' + location.search, location.href)
const ws = new WebSocket('ws' + url.toString().substring(4)) const ws = new WebSocket('ws' + url.toString().substring(4))
@@ -75,7 +100,8 @@
}) })
} }
connect() const media = new URLSearchParams(location.search).get('media')
connect(media || 'video+audio')
</script> </script>
</body> </body>
</html> </html>