我有一个dockerized postgres 9.3.5 OLTP实例，我将更新到9.5.2。而不是关闭它并对文件执行pg_dumpal然后加载它我想启动一个新的docker容器并管道数据库<!DOCTYPE html>
<html>
<head>
<script>
/*
webrtc_polyfill.js by Rob Manson
NOTE: Based on adapter.js by Adam Barth
The MIT License
Copyright (c) 2010-2013 Rob Manson, http://buildAR.com. All rights reserved.
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in
all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
THE SOFTWARE.
*/
var webrtc_capable = true;
var rtc_peer_connection = null;
var rtc_session_description = null;
var get_user_media = null;
var connect_stream_to_src = null;
var stun_server = "stun.l.google.com:19302";
if (navigator.getUserMedia) { // WebRTC 1.0 standard compliant browser
rtc_peer_connection = RTCPeerConnection;
rtc_session_description = RTCSessionDescription;
get_user_media = navigator.getUserMedia.bind(navigator);
connect_stream_to_src = function(media_stream, media_element) {
// https://www.w3.org/Bugs/Public/show_bug.cgi?id=21606
media_element.srcObject = media_stream;
media_element.play();
};
} else if (navigator.mediaDevices.getUserMedia) { // early firefox webrtc implementation
rtc_peer_connection = mozRTCPeerConnection;
rtc_session_description = mozRTCSessionDescription;
get_user_media = navigator.mozGetUserMedia.bind(navigator);
connect_stream_to_src = function(media_stream, media_element) {
media_element.mozSrcObject = media_stream;
media_element.play();
};
stun_server = "74.125.31.127:19302";
} else if (navigator.webkitGetUserMedia) { // early webkit webrtc implementation
rtc_peer_connection = webkitRTCPeerConnection;
rtc_session_description = RTCSessionDescription;
get_user_media = navigator.webkitGetUserMedia.bind(navigator);
connect_stream_to_src = function(media_stream, media_element) {
media_element.src = webkitURL.createObjectURL(media_stream);
};
} else {
alert("This browser does not support WebRTC - visit WebRTC.org for more info");
webrtc_capable = false;
}
</script>
<script>
/*
basic_video_call.js by Rob Manson
The MIT License
Copyright (c) 2010-2013 Rob Manson, http://buildAR.com. All rights reserved.
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in
all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
THE SOFTWARE.
*/
var call_token; // unique token for this call
var signaling_server; // signaling server for this call
var peer_connection; // peer connection object
function start() {
// create the WebRTC peer connection object
peer_connection = new rtc_peer_connection({ // RTCPeerConnection configuration
"iceServers": [ // information about ice servers
{ "url": "stun:"+stun_server }, // stun server info
]
});
// generic handler that sends any ice candidates to the other peer
peer_connection.onicecandidate = function (ice_event) {
if (ice_event.candidate) {
signaling_server.send(
JSON.stringify({
token:call_token,
type: "new_ice_candidate",
candidate: ice_event.candidate ,
})
);
}
};
// display remote video streams when they arrive using local <video> MediaElement
peer_connection.onaddstream = function (event) {
connect_stream_to_src(event.stream, document.getElementById("remote_video"));
// hide placeholder and show remote video
document.getElementById("loading_state").style.display = "none";
document.getElementById("open_call_state").style.display = "block";
};
// setup stream from the local camera
setup_video();
// setup generic connection to the signaling server using the WebSocket API
signaling_server = new WebSocket("ws://localhost:8000");
if (document.location.hash === "" || document.location.hash === undefined) { // you are the Caller
// create the unique token for this call
var token = Math.round(Math.random()*100);
call_token = "#"+token;
// set location.hash to the unique token for this call
document.location.hash = token;
signaling_server.onopen = function() {
// setup caller signal handler
signaling_server.onmessage = caller_signal_handler;
// tell the signaling server you have joined the call
signaling_server.send(
JSON.stringify({
token:call_token,
type:"join",
})
);
}
document.title = "You are the Caller";
document.getElementById("loading_state").innerHTML = "Ready for a call...ask your friend to visit:<br/><br/>"+document.location;
} else { // you have a hash fragment so you must be the Callee
// get the unique token for this call from location.hash
call_token = document.location.hash;
signaling_server.onopen = function() {
// setup caller signal handler
signaling_server.onmessage = callee_signal_handler;
// tell the signaling server you have joined the call
signaling_server.send(
JSON.stringify({
token:call_token,
type:"join",
})
);
// let the caller know you have arrived so they can start the call
signaling_server.send(
JSON.stringify({
token:call_token,
type:"callee_arrived",
})
);
}
document.title = "You are the Callee";
document.getElementById("loading_state").innerHTML = "One moment please...connecting your call...";
}
}
/* functions used above are defined below */
// handler to process new descriptions
function new_description_created(description) {
peer_connection.setLocalDescription(
description,
function () {
signaling_server.send(
JSON.stringify({
token:call_token,
type:"new_description",
sdp:description
})
);
},
log_error
);
}
// handle signals as a caller
function caller_signal_handler(event) {
var signal = JSON.parse(event.data);
if (signal.type === "callee_arrived") {
peer_connection.createOffer(
new_description_created,
log_error
);
} else if (signal.type === "new_ice_candidate") {
peer_connection.addIceCandidate(
new RTCIceCandidate(signal.candidate)
);
} else if (signal.type === "new_description") {
peer_connection.setRemoteDescription(
new rtc_session_description(signal.sdp),
function () {
if (peer_connection.remoteDescription.type == "answer") {
// extend with your own custom answer handling here
}
},
log_error
);
} else {
// extend with your own signal types here
}
}
// handle signals as a callee
function callee_signal_handler(event) {
var signal = JSON.parse(event.data);
if (signal.type === "new_ice_candidate") {
peer_connection.addIceCandidate(
new RTCIceCandidate(signal.candidate)
);
} else if (signal.type === "new_description") {
peer_connection.setRemoteDescription(
new rtc_session_description(signal.sdp),
function () {
if (peer_connection.remoteDescription.type == "offer") {
peer_connection.createAnswer(new_description_created, log_error);
}
},
log_error
);
} else {
// extend with your own signal types here
}
}
// setup stream from the local camera
function setup_video() {
get_user_media(
{
"audio": true, // request access to local microphone
"video": true // request access to local camera
//"video": {mandatory: {minHeight:8, maxHeight:8, minWidth:8, maxWidth:8}}
},
function (local_stream) { // success callback
// display preview from the local camera & microphone using local <video> MediaElement
connect_stream_to_src(local_stream, document.getElementById("local_video"));
// add local camera stream to peer_connection ready to be sent to the remote peer
peer_connection.addStream(local_stream);
},
log_error
);
}
// generic error handler
function log_error(error) {
console.log(error);
}
</script>
<style>
html, body {
padding: 0px;
margin: 0px;
font-family: "Arial","Helvetica",sans-serif;
}
#loading_state {
position: absolute;
top: 45%;
left: 0px;
width: 100%;
font-size: 20px;
text-align: center;
}
#open_call_state {
display: none;
}
#local_video {
position: absolute;
top: 10px;
left: 10px;
width: 160px;
height: 120px;
background: #333333;
}
#remote_video {
position: absolute;
top: 0px;
left: 0px;
width: 1024px;
height: 768px;
background: #999999;
}
</style>
</head>
<body onload="start()">
<div id="loading_state">
loading...
</div>
<div id="open_call_state">
<video id="remote_video"></video>
<video id="local_video"></video>
</div>
</body>
</html>
。我以为我可以将pg_dumpall -h localhost -p [port] -U postgres | psql -h localhost -U postgres -p [port]改为只读模式。这会暂时弄乱我的应用，但至少用户仍然可以postgresql.conf。有没有更好的方法来解决这个问题？在管道数据库时，是否存在放入只读模式的大问题？