diff --git a/README.md b/README.md
index a83b42f..ea078b7 100644
--- a/README.md
+++ b/README.md
@@ -107,18 +107,34 @@ python app.py --fullbody --fullbody_img data/fullbody/img --fullbody_offset_x 10
- ernerf训练第三步torso如果训练的不好，在拼接处会有接缝。可以在上面的命令加上--torso_imgs data/xxx/torso_imgs，torso不用模型推理，直接用训练数据集里的torso图片。这种方式可能头颈处会有些人工痕迹。
### 3.6 webrtc
+#### 3.6.1 p2p模式
+此种模式不需要srs
```
python app.py --transport webrtc
```
用浏览器打开http://serverip:8010/webrtc.html
+#### 3.6.2 通过srs一对多
+启动srs
+```
+export CANDIDATE='<服务器外网ip>'
+docker run --rm --env CANDIDATE=$CANDIDATE \
+ -p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
+ registry.cn-hangzhou.aliyuncs.com/ossrs/srs:5 \
+ objs/srs -c conf/rtc.conf
+```
+然后运行
+```
+python app.py --transport rtcpush --push_url 'http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream'
+```
+用浏览器打开http://serverip:8010/rtcpush.html
## 4. Docker Run
不需要第1步的安装，直接运行。
```
docker run --gpus all -it --network=host --rm registry.cn-hangzhou.aliyuncs.com/lipku/nerfstream:v1.3
```
-srs的运行同2.1
+docker版本已经不是最新代码，可以作为一个空环境，把最新代码拷进去运行。
## 5. Data flow
diff --git a/app.py b/app.py
index b103687..2738379 100644
--- a/app.py
+++ b/app.py
@@ -14,6 +14,7 @@ from threading import Thread,Event
import multiprocessing
from aiohttp import web
+import aiohttp
from aiortc import RTCPeerConnection, RTCSessionDescription
from webrtc import HumanPlayer
@@ -244,6 +245,33 @@ async def on_shutdown(app):
coros = [pc.close() for pc in pcs]
await asyncio.gather(*coros)
pcs.clear()
+
+async def post(url,data):
+ try:
+ async with aiohttp.ClientSession() as session:
+ async with session.post(url,data=data) as response:
+ return await response.text()
+ except aiohttp.ClientError as e:
+ print(f'Error: {e}')
+
+async def run(push_url):
+ pc = RTCPeerConnection()
+ pcs.add(pc)
+
+ @pc.on("connectionstatechange")
+ async def on_connectionstatechange():
+ print("Connection state is %s" % pc.connectionState)
+ if pc.connectionState == "failed":
+ await pc.close()
+ pcs.discard(pc)
+
+ player = HumanPlayer(nerfreal)
+ audio_sender = pc.addTrack(player.audio)
+ video_sender = pc.addTrack(player.video)
+
+ await pc.setLocalDescription(await pc.createOffer())
+ answer = await post(push_url,pc.localDescription.sdp)
+ await pc.setRemoteDescription(RTCSessionDescription(sdp=answer,type='answer'))
##########################################
if __name__ == '__main__':
@@ -344,8 +372,8 @@ if __name__ == '__main__':
# parser.add_argument('--asr_model', type=str, default='facebook/wav2vec2-large-960h-lv60-self')
# parser.add_argument('--asr_model', type=str, default='facebook/hubert-large-ls960-ft')
- parser.add_argument('--transport', type=str, default='rtmp') #rtmp webrtc
- parser.add_argument('--push_url', type=str, default='rtmp://localhost/live/livestream')
+ parser.add_argument('--transport', type=str, default='rtmp') #rtmp webrtc rtcpush
+ parser.add_argument('--push_url', type=str, default='rtmp://localhost/live/livestream') #http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
parser.add_argument('--asr_save_feats', action='store_true')
# audio FPS
@@ -437,6 +465,8 @@ if __name__ == '__main__':
loop.run_until_complete(runner.setup())
site = web.TCPSite(runner, '0.0.0.0', 8010)
loop.run_until_complete(site.start())
+ if opt.transport=='rtcpush':
+ loop.run_until_complete(run(opt.push_url))
loop.run_forever()
Thread(target=run_server, args=(web.AppRunner(appasync),)).start()
diff --git a/asrreal.py b/asrreal.py
index 080c4b2..457feeb 100644
--- a/asrreal.py
+++ b/asrreal.py
@@ -191,7 +191,7 @@ class ASR:
if not self.terminated:
self.frames = self.frames[-(self.stride_left_size + self.stride_right_size):]
- print(f'[INFO] frame_to_text... ')
+ #print(f'[INFO] frame_to_text... ')
#t = time.time()
logits, labels, text = self.__frame_to_text(inputs)
#print(f'-------wav2vec time:{time.time()-t:.4f}s')
diff --git a/web/rtcpush.html b/web/rtcpush.html
new file mode 100644
index 0000000..c2a5c5f
--- /dev/null
+++ b/web/rtcpush.html
@@ -0,0 +1,125 @@
+
+
+
+
+
+ WebRTC webcam
+
+
+
+
+
+
+
+
+
+
+
+
+
Media
+
+
+
+
+
+
+
+
+
+
diff --git a/web/srs.sdk.js b/web/srs.sdk.js
new file mode 100644
index 0000000..2a59788
--- /dev/null
+++ b/web/srs.sdk.js
@@ -0,0 +1,698 @@
+
+//
+// Copyright (c) 2013-2021 Winlin
+//
+// SPDX-License-Identifier: MIT
+//
+
+'use strict';
+
+function SrsError(name, message) {
+ this.name = name;
+ this.message = message;
+ this.stack = (new Error()).stack;
+}
+SrsError.prototype = Object.create(Error.prototype);
+SrsError.prototype.constructor = SrsError;
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-awat-prmise based SRS RTC Publisher.
+function SrsRtcPublisherAsync() {
+ var self = {};
+
+ // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+ self.constraints = {
+ audio: true,
+ video: {
+ width: {ideal: 320, max: 576}
+ }
+ };
+
+ // @see https://github.com/rtcdn/rtcdn-draft
+ // @url The WebRTC url to play with, for example:
+ // webrtc://r.ossrs.net/live/livestream
+ // or specifies the API port:
+ // webrtc://r.ossrs.net:11985/live/livestream
+ // or autostart the publish:
+ // webrtc://r.ossrs.net/live/livestream?autostart=true
+ // or change the app from live to myapp:
+ // webrtc://r.ossrs.net:11985/myapp/livestream
+ // or change the stream from livestream to mystream:
+ // webrtc://r.ossrs.net:11985/live/mystream
+ // or set the api server to myapi.domain.com:
+ // webrtc://myapi.domain.com/live/livestream
+ // or set the candidate(eip) of answer:
+ // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
+ // or force to access https API:
+ // webrtc://r.ossrs.net/live/livestream?schema=https
+ // or use plaintext, without SRTP:
+ // webrtc://r.ossrs.net/live/livestream?encrypt=false
+ // or any other information, will pass-by in the query:
+ // webrtc://r.ossrs.net/live/livestream?vhost=xxx
+ // webrtc://r.ossrs.net/live/livestream?token=xxx
+ self.publish = async function (url) {
+ var conf = self.__internal.prepareUrl(url);
+ self.pc.addTransceiver("audio", {direction: "sendonly"});
+ self.pc.addTransceiver("video", {direction: "sendonly"});
+ //self.pc.addTransceiver("video", {direction: "sendonly"});
+ //self.pc.addTransceiver("audio", {direction: "sendonly"});
+
+ if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+ throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+ }
+ var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ stream.getTracks().forEach(function (track) {
+ self.pc.addTrack(track);
+
+ // Notify about local track when stream is ok.
+ self.ontrack && self.ontrack({track: track});
+ });
+
+ var offer = await self.pc.createOffer();
+ await self.pc.setLocalDescription(offer);
+ var session = await new Promise(function (resolve, reject) {
+ // @see https://github.com/rtcdn/rtcdn-draft
+ var data = {
+ api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
+ clientip: null, sdp: offer.sdp
+ };
+ console.log("Generated offer: ", data);
+
+ const xhr = new XMLHttpRequest();
+ xhr.onload = function() {
+ if (xhr.readyState !== xhr.DONE) return;
+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+ const data = JSON.parse(xhr.responseText);
+ console.log("Got answer: ", data);
+ return data.code ? reject(xhr) : resolve(data);
+ }
+ xhr.open('POST', conf.apiUrl, true);
+ xhr.setRequestHeader('Content-type', 'application/json');
+ xhr.send(JSON.stringify(data));
+ });
+ await self.pc.setRemoteDescription(
+ new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+ );
+ session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+ return session;
+ };
+
+ // Close the publisher.
+ self.close = function () {
+ self.pc && self.pc.close();
+ self.pc = null;
+ };
+
+ // The callback when got local stream.
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ self.ontrack = function (event) {
+ // Add track to stream of SDK.
+ self.stream.addTrack(event.track);
+ };
+
+ // Internal APIs.
+ self.__internal = {
+ defaultPath: '/rtc/v1/publish/',
+ prepareUrl: function (webrtcUrl) {
+ var urlObject = self.__internal.parse(webrtcUrl);
+
+ // If user specifies the schema, use it as API schema.
+ var schema = urlObject.user_query.schema;
+ schema = schema ? schema + ':' : window.location.protocol;
+
+ var port = urlObject.port || 1985;
+ if (schema === 'https:') {
+ port = urlObject.port || 443;
+ }
+
+ // @see https://github.com/rtcdn/rtcdn-draft
+ var api = urlObject.user_query.play || self.__internal.defaultPath;
+ if (api.lastIndexOf('/') !== api.length - 1) {
+ api += '/';
+ }
+
+ var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+ for (var key in urlObject.user_query) {
+ if (key !== 'api' && key !== 'play') {
+ apiUrl += '&' + key + '=' + urlObject.user_query[key];
+ }
+ }
+ // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
+ apiUrl = apiUrl.replace(api + '&', api + '?');
+
+ var streamUrl = urlObject.url;
+
+ return {
+ apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
+ tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+ };
+ },
+ parse: function (url) {
+ // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+ var a = document.createElement("a");
+ a.href = url.replace("rtmp://", "http://")
+ .replace("webrtc://", "http://")
+ .replace("rtc://", "http://");
+
+ var vhost = a.hostname;
+ var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+ var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+ // parse the vhost in the params of app, that srs supports.
+ app = app.replace("...vhost...", "?vhost=");
+ if (app.indexOf("?") >= 0) {
+ var params = app.slice(app.indexOf("?"));
+ app = app.slice(0, app.indexOf("?"));
+
+ if (params.indexOf("vhost=") > 0) {
+ vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+ if (vhost.indexOf("&") > 0) {
+ vhost = vhost.slice(0, vhost.indexOf("&"));
+ }
+ }
+ }
+
+ // when vhost equals to server, and server is ip,
+ // the vhost is __defaultVhost__
+ if (a.hostname === vhost) {
+ var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+ if (re.test(a.hostname)) {
+ vhost = "__defaultVhost__";
+ }
+ }
+
+ // parse the schema
+ var schema = "rtmp";
+ if (url.indexOf("://") > 0) {
+ schema = url.slice(0, url.indexOf("://"));
+ }
+
+ var port = a.port;
+ if (!port) {
+ // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+ if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+ port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+ }
+
+ // Guess by schema.
+ if (schema === 'http') {
+ port = 80;
+ } else if (schema === 'https') {
+ port = 443;
+ } else if (schema === 'rtmp') {
+ port = 1935;
+ }
+ }
+
+ var ret = {
+ url: url,
+ schema: schema,
+ server: a.hostname, port: port,
+ vhost: vhost, app: app, stream: stream
+ };
+ self.__internal.fill_query(a.search, ret);
+
+ // For webrtc API, we use 443 if page is https, or schema specified it.
+ if (!ret.port) {
+ if (schema === 'webrtc' || schema === 'rtc') {
+ if (ret.user_query.schema === 'https') {
+ ret.port = 443;
+ } else if (window.location.href.indexOf('https://') === 0) {
+ ret.port = 443;
+ } else {
+ // For WebRTC, SRS use 1985 as default API port.
+ ret.port = 1985;
+ }
+ }
+ }
+
+ return ret;
+ },
+ fill_query: function (query_string, obj) {
+ // pure user query object.
+ obj.user_query = {};
+
+ if (query_string.length === 0) {
+ return;
+ }
+
+ // split again for angularjs.
+ if (query_string.indexOf("?") >= 0) {
+ query_string = query_string.split("?")[1];
+ }
+
+ var queries = query_string.split("&");
+ for (var i = 0; i < queries.length; i++) {
+ var elem = queries[i];
+
+ var query = elem.split("=");
+ obj[query[0]] = query[1];
+ obj.user_query[query[0]] = query[1];
+ }
+
+ // alias domain for vhost.
+ if (obj.domain) {
+ obj.vhost = obj.domain;
+ }
+ }
+ };
+
+ self.pc = new RTCPeerConnection(null);
+
+ // To keep api consistent between player and publisher.
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ // @see https://webrtc.org/getting-started/media-devices
+ self.stream = new MediaStream();
+
+ return self;
+}
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-await-promise based SRS RTC Player.
+function SrsRtcPlayerAsync() {
+ var self = {};
+
+ // @see https://github.com/rtcdn/rtcdn-draft
+ // @url The WebRTC url to play with, for example:
+ // webrtc://r.ossrs.net/live/livestream
+ // or specifies the API port:
+ // webrtc://r.ossrs.net:11985/live/livestream
+ // webrtc://r.ossrs.net:80/live/livestream
+ // or autostart the play:
+ // webrtc://r.ossrs.net/live/livestream?autostart=true
+ // or change the app from live to myapp:
+ // webrtc://r.ossrs.net:11985/myapp/livestream
+ // or change the stream from livestream to mystream:
+ // webrtc://r.ossrs.net:11985/live/mystream
+ // or set the api server to myapi.domain.com:
+ // webrtc://myapi.domain.com/live/livestream
+ // or set the candidate(eip) of answer:
+ // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
+ // or force to access https API:
+ // webrtc://r.ossrs.net/live/livestream?schema=https
+ // or use plaintext, without SRTP:
+ // webrtc://r.ossrs.net/live/livestream?encrypt=false
+ // or any other information, will pass-by in the query:
+ // webrtc://r.ossrs.net/live/livestream?vhost=xxx
+ // webrtc://r.ossrs.net/live/livestream?token=xxx
+ self.play = async function(url) {
+ var conf = self.__internal.prepareUrl(url);
+ self.pc.addTransceiver("audio", {direction: "recvonly"});
+ self.pc.addTransceiver("video", {direction: "recvonly"});
+ //self.pc.addTransceiver("video", {direction: "recvonly"});
+ //self.pc.addTransceiver("audio", {direction: "recvonly"});
+
+ var offer = await self.pc.createOffer();
+ await self.pc.setLocalDescription(offer);
+ var session = await new Promise(function(resolve, reject) {
+ // @see https://github.com/rtcdn/rtcdn-draft
+ var data = {
+ api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
+ clientip: null, sdp: offer.sdp
+ };
+ console.log("Generated offer: ", data);
+
+ const xhr = new XMLHttpRequest();
+ xhr.onload = function() {
+ if (xhr.readyState !== xhr.DONE) return;
+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+ const data = JSON.parse(xhr.responseText);
+ console.log("Got answer: ", data);
+ return data.code ? reject(xhr) : resolve(data);
+ }
+ xhr.open('POST', conf.apiUrl, true);
+ xhr.setRequestHeader('Content-type', 'application/json');
+ xhr.send(JSON.stringify(data));
+ });
+ await self.pc.setRemoteDescription(
+ new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+ );
+ session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+ return session;
+ };
+
+ // Close the player.
+ self.close = function() {
+ self.pc && self.pc.close();
+ self.pc = null;
+ };
+
+ // The callback when got remote track.
+ // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
+ self.ontrack = function (event) {
+ // https://webrtc.org/getting-started/remote-streams
+ self.stream.addTrack(event.track);
+ };
+
+ // Internal APIs.
+ self.__internal = {
+ defaultPath: '/rtc/v1/play/',
+ prepareUrl: function (webrtcUrl) {
+ var urlObject = self.__internal.parse(webrtcUrl);
+
+ // If user specifies the schema, use it as API schema.
+ var schema = urlObject.user_query.schema;
+ schema = schema ? schema + ':' : window.location.protocol;
+
+ var port = urlObject.port || 1985;
+ if (schema === 'https:') {
+ port = urlObject.port || 443;
+ }
+
+ // @see https://github.com/rtcdn/rtcdn-draft
+ var api = urlObject.user_query.play || self.__internal.defaultPath;
+ if (api.lastIndexOf('/') !== api.length - 1) {
+ api += '/';
+ }
+
+ var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+ for (var key in urlObject.user_query) {
+ if (key !== 'api' && key !== 'play') {
+ apiUrl += '&' + key + '=' + urlObject.user_query[key];
+ }
+ }
+ // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
+ apiUrl = apiUrl.replace(api + '&', api + '?');
+
+ var streamUrl = urlObject.url;
+
+ return {
+ apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
+ tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+ };
+ },
+ parse: function (url) {
+ // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+ var a = document.createElement("a");
+ a.href = url.replace("rtmp://", "http://")
+ .replace("webrtc://", "http://")
+ .replace("rtc://", "http://");
+
+ var vhost = a.hostname;
+ var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+ var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+ // parse the vhost in the params of app, that srs supports.
+ app = app.replace("...vhost...", "?vhost=");
+ if (app.indexOf("?") >= 0) {
+ var params = app.slice(app.indexOf("?"));
+ app = app.slice(0, app.indexOf("?"));
+
+ if (params.indexOf("vhost=") > 0) {
+ vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+ if (vhost.indexOf("&") > 0) {
+ vhost = vhost.slice(0, vhost.indexOf("&"));
+ }
+ }
+ }
+
+ // when vhost equals to server, and server is ip,
+ // the vhost is __defaultVhost__
+ if (a.hostname === vhost) {
+ var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+ if (re.test(a.hostname)) {
+ vhost = "__defaultVhost__";
+ }
+ }
+
+ // parse the schema
+ var schema = "rtmp";
+ if (url.indexOf("://") > 0) {
+ schema = url.slice(0, url.indexOf("://"));
+ }
+
+ var port = a.port;
+ if (!port) {
+ // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+ if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+ port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+ }
+
+ // Guess by schema.
+ if (schema === 'http') {
+ port = 80;
+ } else if (schema === 'https') {
+ port = 443;
+ } else if (schema === 'rtmp') {
+ port = 1935;
+ }
+ }
+
+ var ret = {
+ url: url,
+ schema: schema,
+ server: a.hostname, port: port,
+ vhost: vhost, app: app, stream: stream
+ };
+ self.__internal.fill_query(a.search, ret);
+
+ // For webrtc API, we use 443 if page is https, or schema specified it.
+ if (!ret.port) {
+ if (schema === 'webrtc' || schema === 'rtc') {
+ if (ret.user_query.schema === 'https') {
+ ret.port = 443;
+ } else if (window.location.href.indexOf('https://') === 0) {
+ ret.port = 443;
+ } else {
+ // For WebRTC, SRS use 1985 as default API port.
+ ret.port = 1985;
+ }
+ }
+ }
+
+ return ret;
+ },
+ fill_query: function (query_string, obj) {
+ // pure user query object.
+ obj.user_query = {};
+
+ if (query_string.length === 0) {
+ return;
+ }
+
+ // split again for angularjs.
+ if (query_string.indexOf("?") >= 0) {
+ query_string = query_string.split("?")[1];
+ }
+
+ var queries = query_string.split("&");
+ for (var i = 0; i < queries.length; i++) {
+ var elem = queries[i];
+
+ var query = elem.split("=");
+ obj[query[0]] = query[1];
+ obj.user_query[query[0]] = query[1];
+ }
+
+ // alias domain for vhost.
+ if (obj.domain) {
+ obj.vhost = obj.domain;
+ }
+ }
+ };
+
+ self.pc = new RTCPeerConnection(null);
+
+ // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
+ self.stream = new MediaStream();
+
+ // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+ self.pc.ontrack = function(event) {
+ if (self.ontrack) {
+ self.ontrack(event);
+ }
+ };
+
+ return self;
+}
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-awat-prmise based SRS RTC Publisher by WHIP.
+function SrsRtcWhipWhepAsync() {
+ var self = {};
+
+ // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+ self.constraints = {
+ audio: true,
+ video: {
+ width: {ideal: 320, max: 576}
+ }
+ };
+
+ // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+ // @url The WebRTC url to publish with, for example:
+ // http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
+ // @options The options to control playing, supports:
+ // videoOnly: boolean, whether only play video, default to false.
+ // audioOnly: boolean, whether only play audio, default to false.
+ self.publish = async function (url, options) {
+ if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
+ if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
+
+ if (!options?.videoOnly) {
+ self.pc.addTransceiver("audio", {direction: "sendonly"});
+ } else {
+ self.constraints.audio = false;
+ }
+
+ if (!options?.audioOnly) {
+ self.pc.addTransceiver("video", {direction: "sendonly"});
+ } else {
+ self.constraints.video = false;
+ }
+
+ if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+ throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+ }
+ var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ stream.getTracks().forEach(function (track) {
+ self.pc.addTrack(track);
+
+ // Notify about local track when stream is ok.
+ self.ontrack && self.ontrack({track: track});
+ });
+
+ var offer = await self.pc.createOffer();
+ await self.pc.setLocalDescription(offer);
+ const answer = await new Promise(function (resolve, reject) {
+ console.log(`Generated offer: ${offer.sdp}`);
+
+ const xhr = new XMLHttpRequest();
+ xhr.onload = function() {
+ if (xhr.readyState !== xhr.DONE) return;
+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+ const data = xhr.responseText;
+ console.log("Got answer: ", data);
+ return data.code ? reject(xhr) : resolve(data);
+ }
+ xhr.open('POST', url, true);
+ xhr.setRequestHeader('Content-type', 'application/sdp');
+ xhr.send(offer.sdp);
+ });
+ await self.pc.setRemoteDescription(
+ new RTCSessionDescription({type: 'answer', sdp: answer})
+ );
+
+ return self.__internal.parseId(url, offer.sdp, answer);
+ };
+
+ // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+ // @url The WebRTC url to play with, for example:
+ // http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
+ // @options The options to control playing, supports:
+ // videoOnly: boolean, whether only play video, default to false.
+ // audioOnly: boolean, whether only play audio, default to false.
+ self.play = async function(url, options) {
+ if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
+ if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
+
+ if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"});
+ if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"});
+
+ var offer = await self.pc.createOffer();
+ await self.pc.setLocalDescription(offer);
+ const answer = await new Promise(function(resolve, reject) {
+ console.log(`Generated offer: ${offer.sdp}`);
+
+ const xhr = new XMLHttpRequest();
+ xhr.onload = function() {
+ if (xhr.readyState !== xhr.DONE) return;
+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+ const data = xhr.responseText;
+ console.log("Got answer: ", data);
+ return data.code ? reject(xhr) : resolve(data);
+ }
+ xhr.open('POST', url, true);
+ xhr.setRequestHeader('Content-type', 'application/sdp');
+ xhr.send(offer.sdp);
+ });
+ await self.pc.setRemoteDescription(
+ new RTCSessionDescription({type: 'answer', sdp: answer})
+ );
+
+ return self.__internal.parseId(url, offer.sdp, answer);
+ };
+
+ // Close the publisher.
+ self.close = function () {
+ self.pc && self.pc.close();
+ self.pc = null;
+ };
+
+ // The callback when got local stream.
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ self.ontrack = function (event) {
+ // Add track to stream of SDK.
+ self.stream.addTrack(event.track);
+ };
+
+ self.pc = new RTCPeerConnection(null);
+
+ // To keep api consistent between player and publisher.
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ // @see https://webrtc.org/getting-started/media-devices
+ self.stream = new MediaStream();
+
+ // Internal APIs.
+ self.__internal = {
+ parseId: (url, offer, answer) => {
+ let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+ sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
+ sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+ sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
+
+ const a = document.createElement("a");
+ a.href = url;
+ return {
+ sessionid: sessionid, // Should be ice-ufrag of answer:offer.
+ simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
+ };
+ },
+ };
+
+ // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+ self.pc.ontrack = function(event) {
+ if (self.ontrack) {
+ self.ontrack(event);
+ }
+ };
+
+ return self;
+}
+
+// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+function SrsRtcFormatSenders(senders, kind) {
+ var codecs = [];
+ senders.forEach(function (sender) {
+ var params = sender.getParameters();
+ params && params.codecs && params.codecs.forEach(function(c) {
+ if (kind && sender.track.kind !== kind) {
+ return;
+ }
+
+ if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
+ return;
+ }
+
+ var s = '';
+
+ s += c.mimeType.replace('audio/', '').replace('video/', '');
+ s += ', ' + c.clockRate + 'HZ';
+ if (sender.track.kind === "audio") {
+ s += ', channels: ' + c.channels;
+ }
+ s += ', pt: ' + c.payloadType;
+
+ codecs.push(s);
+ });
+ });
+ return codecs.join(", ");
+}
+