add api to human
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lipku
parent
8c012c5ab8
commit
3e702b8a12
53
app.py
53
app.py
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@ -206,6 +206,37 @@ def chat_socket(ws):
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#####webrtc###############################
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pcs = set()
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async def txt_to_audio_async(text_):
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if tts_type == "edgetts":
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voicename = "zh-CN-YunxiaNeural"
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text = text_
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t = time.time()
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#asyncio.get_event_loop().run_until_complete(main(voicename,text,nerfreal))
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await main(voicename,text,nerfreal)
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print(f'-------edge tts time:{time.time()-t:.4f}s')
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elif tts_type == "gpt-sovits": #gpt_sovits
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stream_tts(
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gpt_sovits(
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text_,
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app.config['CHARACTER'], #"test", #character
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"zh", #en args.language,
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app.config['TTS_SERVER'], #"http://127.0.0.1:5000", #args.server_url,
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app.config['EMOTION'], #emotion
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),
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nerfreal
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)
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else: #xtts
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stream_tts(
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xtts(
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text_,
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gspeaker,
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"zh-cn", #en args.language,
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app.config['TTS_SERVER'], #"http://localhost:9000", #args.server_url,
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"20" #args.stream_chunk_size
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),
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nerfreal
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)
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#@app.route('/offer', methods=['POST'])
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async def offer(request):
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params = await request.json()
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@ -239,6 +270,21 @@ async def offer(request):
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),
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)
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async def human(request):
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params = await request.json()
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if params['type']=='echo':
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await txt_to_audio_async(params['text'])
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elif params['type']=='chat':
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res=llm_response(params['text'])
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await txt_to_audio_async(res)
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "data":"ok"}
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),
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)
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async def on_shutdown(app):
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# close peer connections
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@ -400,7 +446,7 @@ if __name__ == '__main__':
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parser.add_argument('--CHARACTER', type=str, default='test')
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parser.add_argument('--EMOTION', type=str, default='default')
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parser.add_argument('--listenport', type=int, default=8000)
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parser.add_argument('--listenport', type=int, default=8010)
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opt = parser.parse_args()
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app.config.from_object(opt)
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@ -463,13 +509,14 @@ if __name__ == '__main__':
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appasync = web.Application()
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appasync.on_shutdown.append(on_shutdown)
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appasync.router.add_post("/offer", offer)
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appasync.router.add_post("/human", human)
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appasync.router.add_static('/',path='web')
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def run_server(runner):
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loop = asyncio.new_event_loop()
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asyncio.set_event_loop(loop)
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loop.run_until_complete(runner.setup())
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site = web.TCPSite(runner, '0.0.0.0', 8010)
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site = web.TCPSite(runner, '0.0.0.0', opt.listenport)
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loop.run_until_complete(site.start())
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if opt.transport=='rtcpush':
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loop.run_until_complete(run(opt.push_url))
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@ -479,7 +526,7 @@ if __name__ == '__main__':
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print('start websocket server')
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#app.on_shutdown.append(on_shutdown)
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#app.router.add_post("/offer", offer)
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server = pywsgi.WSGIServer(('0.0.0.0', opt.listenport), app, handler_class=WebSocketHandler)
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server = pywsgi.WSGIServer(('0.0.0.0', 8000), app, handler_class=WebSocketHandler)
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server.serve_forever()
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@ -0,0 +1,135 @@
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<!DOCTYPE html>
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<html>
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<head>
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<meta charset="UTF-8"/>
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<meta name="viewport" content="width=device-width, initial-scale=1.0" />
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<title>WebRTC webcam</title>
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<style>
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button {
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padding: 8px 16px;
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}
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video {
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width: 100%;
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}
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.option {
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margin-bottom: 8px;
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}
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#media {
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max-width: 1280px;
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}
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</style>
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</head>
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<body>
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<div class="option">
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<input id="use-stun" type="checkbox"/>
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<label for="use-stun">Use STUN server</label>
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</div>
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<button class="btn btn-primary" id="btn_play">Start</button>
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<form class="form-inline" id="echo-form">
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<div class="form-group">
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<p>input text</p>
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<textarea cols="2" rows="3" style="width:600px;height:50px;" class="form-control" id="message">test</textarea>
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</div>
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<button type="submit" class="btn btn-default">Send</button>
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</form>
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<div id="media">
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<h2>Media</h2>
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<video id="rtc_media_player" style="width:600px;" controls autoplay></video>
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</div>
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<script src="srs.sdk.js"></script>
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<script type="text/javascript" src="http://cdn.sockjs.org/sockjs-0.3.4.js"></script>
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<script type="text/javascript" src="https://ajax.aspnetcdn.com/ajax/jquery/jquery-2.1.1.min.js"></script>
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</body>
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<script type="text/javascript" charset="utf-8">
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$(document).ready(function() {
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// var host = window.location.hostname
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// var ws = new WebSocket("ws://"+host+":8000/humanecho");
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// //document.getElementsByTagName("video")[0].setAttribute("src", aa["video"]);
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// ws.onopen = function() {
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// console.log('Connected');
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// };
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// ws.onmessage = function(e) {
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// console.log('Received: ' + e.data);
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// data = e
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// var vid = JSON.parse(data.data);
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// console.log(typeof(vid),vid)
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// //document.getElementsByTagName("video")[0].setAttribute("src", vid["video"]);
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// };
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// ws.onclose = function(e) {
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// console.log('Closed');
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// };
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$('#echo-form').on('submit', function(e) {
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e.preventDefault();
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var message = $('#message').val();
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console.log('Sending: ' + message);
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fetch('/human', {
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body: JSON.stringify({
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text: message,
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type: 'echo',
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}),
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headers: {
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'Content-Type': 'application/json'
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},
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method: 'POST'
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});
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//ws.send(message);
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$('#message').val('');
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});
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});
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$(function(){
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var sdk = null; // Global handler to do cleanup when republishing.
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var startPlay = function() {
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$('#rtc_media_player').show();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcWhipWhepAsync();
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// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
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// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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var host = window.location.hostname
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = "http://"+host+":1985/rtc/v1/whep/?app=live&stream=livestream"
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sdk.play(url).then(function(session){
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//$('#sessionid').html(session.sessionid);
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//$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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// var query = parse_query_string();
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// srs_init_whep("#txt_url", query);
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$("#btn_play").click(startPlay);
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// Never play util windows loaded @see https://github.com/ossrs/srs/issues/2732
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// if (query.autostart === 'true') {
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// $('#rtc_media_player').prop('muted', true);
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// console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
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// 'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
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// window.addEventListener("load", function(){ startPlay(); });
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// }
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});
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</script>
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</html>
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@ -0,0 +1,93 @@
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<!DOCTYPE html>
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<html>
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<head>
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<meta charset="UTF-8"/>
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<meta name="viewport" content="width=device-width, initial-scale=1.0" />
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<title>WebRTC webcam</title>
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<style>
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button {
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padding: 8px 16px;
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}
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video {
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width: 100%;
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}
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.option {
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margin-bottom: 8px;
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}
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#media {
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max-width: 1280px;
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}
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</style>
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</head>
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<body>
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<div class="option">
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<input id="use-stun" type="checkbox"/>
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<label for="use-stun">Use STUN server</label>
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</div>
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<button id="start" onclick="start()">Start</button>
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<button id="stop" style="display: none" onclick="stop()">Stop</button>
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<form class="form-inline" id="echo-form">
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<div class="form-group">
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<p>input text</p>
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<textarea cols="2" rows="3" style="width:600px;height:50px;" class="form-control" id="message">test</textarea>
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</div>
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<button type="submit" class="btn btn-default">Send</button>
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</form>
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<div id="media">
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<h2>Media</h2>
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<audio id="audio" autoplay="true"></audio>
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<video id="video" style="width:600px;" autoplay="true" playsinline="true"></video>
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</div>
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<script src="client.js"></script>
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<script type="text/javascript" src="http://cdn.sockjs.org/sockjs-0.3.4.js"></script>
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<script type="text/javascript" src="https://ajax.aspnetcdn.com/ajax/jquery/jquery-2.1.1.min.js"></script>
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</body>
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<script type="text/javascript" charset="utf-8">
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$(document).ready(function() {
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// var host = window.location.hostname
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// var ws = new WebSocket("ws://"+host+":8000/humanecho");
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// //document.getElementsByTagName("video")[0].setAttribute("src", aa["video"]);
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// ws.onopen = function() {
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// console.log('Connected');
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// };
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// ws.onmessage = function(e) {
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// console.log('Received: ' + e.data);
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// data = e
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// var vid = JSON.parse(data.data);
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// console.log(typeof(vid),vid)
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// //document.getElementsByTagName("video")[0].setAttribute("src", vid["video"]);
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// };
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// ws.onclose = function(e) {
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// console.log('Closed');
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// };
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$('#echo-form').on('submit', function(e) {
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e.preventDefault();
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var message = $('#message').val();
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console.log('Sending: ' + message);
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fetch('/human', {
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body: JSON.stringify({
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text: message,
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type: 'echo',
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}),
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headers: {
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'Content-Type': 'application/json'
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},
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method: 'POST'
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});
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//ws.send(message);
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$('#message').val('');
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});
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});
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</script>
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</html>
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