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Removed the obsolete sanity check and added new test HTML files.

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Review URL: https://webrtc-codereview.appspot.com/630004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2349 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org 2012-06-04 10:06:52 +00:00
parent ab12990b1b
commit f1d6e0a65b
13 changed files with 319 additions and 847 deletions
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8
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================================================================
WEBRTC MANUAL TESTS
================================================================
You will need to serve these files off some kind of web server. Currently,
GetUserMedia does not work when run off a file:// URL.
Contact person: phoglund@webrtc.org

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Single Local Preview (Video and Audio)</title>
<script type="text/javascript">
function requestVideoAndAudio() {
navigator.webkitGetUserMedia({video: true, audio: true},
getUserMediaOkCallback,
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream) {
var streamUrl = webkitURL.createObjectURL(stream);
document.getElementById("view1").src = streamUrl;
//document.getElementById("audio1").src = streamUrl;
}
</script>
</head>
<body onload="requestVideoAndAudio();">
<table border="0">
<tr>
<td>Local Preview</td>
</tr>
<tr>
<td><video width="320" height="240" id="view1"
autoplay="autoplay"></video></td>
<td><audio id="audio1" autoplay="autoplay"></audio></td>
</tr>
</table>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>IFRAME Single Local Preview (Video Only)</title>
</head>
<body>
<iframe width="100%" height="100%" src="single-video.html"></iframe>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Multiple Local Preview (Audio Only)</title>
<script type="text/javascript">
function requestAudio() {
navigator.webkitGetUserMedia({video: false, audio: true},
getUserMediaOkCallback,
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream) {
var streamUrl = webkitURL.createObjectURL(stream);
for (var i = 1; i <= 10; i++) {
document.getElementById("audio" + i).src = streamUrl;
}
}
</script>
</head>
<body onload="requestAudio();">
<table border="0">
<tr>
<td>Sound test</td>
</tr>
<tr>
<td><audio id="audio1" autoplay="autoplay"></audio></td>
<td><audio id="audio2" autoplay="autoplay"></audio></td>
<td><audio id="audio3" autoplay="autoplay"></audio></td>
<td><audio id="audio4" autoplay="autoplay"></audio></td>
<td><audio id="audio5" autoplay="autoplay"></audio></td>
<td><audio id="audio6" autoplay="autoplay"></audio></td>
<td><audio id="audio7" autoplay="autoplay"></audio></td>
<td><audio id="audio8" autoplay="autoplay"></audio></td>
<td><audio id="audio9" autoplay="autoplay"></audio></td>
<td><audio id="audio10" autoplay="autoplay"></audio></td>
</tr>
</table>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Multiple Local Preview (Video Only)</title>
<script type="text/javascript">
function requestVideo() {
navigator.webkitGetUserMedia({video: true, audio: false},
getUserMediaOkCallback,
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream) {
var streamUrl = webkitURL.createObjectURL(stream);
for (var i = 1; i <= 10; i++) {
document.getElementById("view" + i).src = streamUrl;
}
}
</script>
</head>
<body onload="requestVideo();">
<table border="0">
<tr>
<td>Local Preview</td>
</tr>
<tr>
<td><video width="320" height="240" id="view1"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view2"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view3"
autoplay="autoplay"></video></td>
</tr>
<tr>
<td><video width="320" height="240" id="view4"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view5"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view6"
autoplay="autoplay"></video></td>
</tr>
<tr>
<td><video width="320" height="240" id="view7"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view8"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view9"
autoplay="autoplay"></video></td>
</tr>
<tr>
<td><video width="320" height="240" id="view10"
autoplay="autoplay"></video></td>
</tr>
</table>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Single Local Preview (Audio Only)</title>
<script type="text/javascript">
function requestAudio() {
navigator.webkitGetUserMedia({video: false, audio: true},
getUserMediaOkCallback,
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream) {
var streamUrl = webkitURL.createObjectURL(stream);
document.getElementById("audio1").src = streamUrl;
}
</script>
</head>
<body onload="requestAudio();">
<table border="0">
<tr>
<td>Sound test</td>
</tr>
<tr>
<td><audio id="audio1" autoplay="autoplay"></audio></td>
</tr>
</table>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Single Local Preview (Video Only)</title>
<script type="text/javascript">
function requestVideo() {
navigator.webkitGetUserMedia({video: true, audio: false},
getUserMediaOkCallback,
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream) {
var streamUrl = webkitURL.createObjectURL(stream);
document.getElementById("view1").src = streamUrl;
}
</script>
</head>
<body onload="requestVideo();">
<table border="0">
<tr>
<td>Local Preview</td>
</tr>
<tr>
<td><video width="320" height="240" id="view1"
autoplay="autoplay"></video></td>
</tr>
</table>
</body>
</html>

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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>Single Local Preview (Video Only)</title>
<script type="text/javascript">
function requestVideo(target) {
navigator.webkitGetUserMedia({video: true, audio: false},
function(stream) {
getUserMediaOkCallback(stream, target);
},
getUserMediaFailedCallback);
}
function getUserMediaFailedCallback(error) {
alert("User media request denied with error code " + error.code);
}
function getUserMediaOkCallback(stream, target) {
var streamUrl = webkitURL.createObjectURL(stream);
document.getElementById(target).src = streamUrl;
}
</script>
</head>
<body onload="requestVideo('view1'); requestVideo('view2');">
<table border="0">
<tr>
<td>Local Preview</td>
</tr>
<tr>
<td><video width="320" height="240" id="view1"
autoplay="autoplay"></video></td>
<td><video width="320" height="240" id="view2"
autoplay="autoplay"></video></td>
</tr>
</table>
</body>
</html>

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This test will pop up a browser with two tabs, and will run video from
one tab to the other tab, across the local network interface.
In order to run this test, you need to have:
- an installed lighttpd in your $PATH
- a built peerconnection_server
- a sufficiently new google-chrome binary with WebRTC support.
To build peerconnection_server, refer to peerconnection_server/README. This
script will assume you have a directory structure like so:
- webrtc/trunk/
- peerconnection/trunk
That is, the script assumes it will find a peerconnection_server binary in
../../../../peerconnection/out/Debug/ relative to the sanity check
directory. If this doesn't suit you, feel free to change the script locally.
You need to run the script from the test/sanity_check directory. The first time
you run it, you will be asked to choose a search engine. Additionally, you will
have to let Chrome access your webcam by accepting the notice at the top.
This script uses the webrtc_test.html code, which is described by its own
README in the www/html directory.

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server.document-root = "./www/html/"
server.port = 3000
mimetype.assign = (
".html" => "text/html"
)

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#!/bin/bash
#
# Run a test with the WebRTC Chromium build.
# Should work on any machine with a camera.
#
# Pass the argument --xvfb-and-screenshots to run in a xvfb environment and
# capture screenshots from inside the virtual environment. If you don't what
# that is, never mind and run this script without arguments.
#
# Pass the argument --chrome-binary <binary> to provide a custom chrome binary.
#
# Method:
# - Start peerconnection_server.
# - Start serving webrtc_test.html off a local web server.
# - Start 2 browser tabs.
# - Tab 1 connects to the server.
# - Tab 2 connects to the server and calls tab 1.
# - Both tabs will capture the webcam you have on the system and start
# sending to each other.
#
set -e
if [ ! -x run_sanity_check ]; then
echo "Error: This script must run from its own directory"
exit 1
fi
# Tweakable parameters:
URLBASE=localhost:3000
CALLEE="$URLBASE/webrtc_test.html?name=test1&autoconnect=yes"
CALLER="$URLBASE/webrtc_test.html?name=test2&autocall=test1"
FLAGS="--enable-media-stream"
CHROME_BINARY=google-chrome
# Parse parameters:
while (( "$#" )); do
if [ "$1" == "--xvfb-and-screenshots" ]; then
RUN_WITH_XVFB_AND_SCREENSHOTS=1
fi
if [ "$1" == "--chrome-binary" ]; then
shift
CHROME_BINARY=$1
fi
shift
done
run_chrome_for_user() {
# Run the binary straight up and proceed with cleanup when the user closes
# the browser.
${CHROME_BINARY} $CALLEE $CALLER $FLAGS --user-data-dir=temp/user1
}
run_chrome_in_xvfb_with_screenshots() {
# Run chrome in a virtual window environment and capture screenshots.
# It will run for 30 seconds and then exit.
Xvfb :17 -screen 0 1024x768x24 &
XVFB=$!
DISPLAY=:17 ${CHROME_BINARY} $CALLEE $CALLER $FLAGS --user-data-dir=temp/user1 &
CHROME=$!
sleep 2
for (( i = 0; i < 6; i++ )) do
FILENAME="IMAGE${i}.png"
DISPLAY=:17 import -window root $FILENAME
sleep 5
done
kill $CHROME || echo "No Chrome process."
kill $XVFB || echo "No XVFB process."
}
PROJECT_ROOT=../..
if which lighttpd; then
LOCAL_WEB_SERVER_BINARY=$(which lighttpd)
else
echo "Error: You must install lighttpd first (sudo apt-get install lighttpd)"
exit 1
fi
SERVER_BINARY=${PROJECT_ROOT}/../../peerconnection/trunk/out/Debug/peerconnection_server
if [ ! -e "$SERVER_BINARY" ]; then
echo "Error: You must build peerconnection_server first."
echo "I expected to find a binary at $SERVER_BINARY but found none."
exit 1
fi
# Launch the web server and make it serve the local www/html directory
${LOCAL_WEB_SERVER_BINARY} -D -f lighttpd.conf &
LOCAL_WEB_SERVER=$!
${SERVER_BINARY} &
SERVER=$!
echo "Started server as $SERVER"
# Launch a browser with two tabs:
if [ -z "$RUN_WITH_XVFB_AND_SCREENSHOTS" ]; then
run_chrome_for_user
else
run_chrome_in_xvfb_with_screenshots
fi
echo "Test finished, cleaning up"
kill $SERVER || echo "No peerconnection_server"
kill $LOCAL_WEB_SERVER || echo "No local web server"

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webrtc_test.html is a simple functional test for a WebRTC enabled browser. It
has only been tested with Chrome, and is most likely only working with Chrome at
the moment. The following instructions are in part Chrome specific.
If you want a quick, automated test, have a look at the test/sanity_check script
which uses this page to quickly launch a call test on Linux.
Table of contents:
* PREREQUISITES
* SINGLE FAKE CLIENT IN LOOPBACK MODE
* TWO CLIENTS CALLING EACH OTHER
* TWO AUTOCALLING CLIENTS
* PREREQUISITES
The following is necessary to run the test:
- A WebRTC enabled Chrome binary. (Available in dev or canary channel, version
18.0.1008 or newer.)
- A peerconnection_server binary. See peerconnection/README for more information
on how to do this.
Note: The web page must normally be on a web server to be able to access the
camera for security reasons.
See http://blog.chromium.org/2008/12/security-in-depth-local-web-pages.html
for more details on this topic. This can be overridden with the flag
--allow-file-access-from-files, in which case running it over the file://
URI scheme works.
* SINGLE FAKE CLIENT IN LOOPBACK MODE
1. Start peerconnection_server.
2. Start the WebRTC Chrome build:
$ <path_to_chrome_binary>/chrome --enable-media-stream
The --enable-media-stream flag is required for the time being.
3. Open the server test page, ensure loopback is enabled, choose a name (for
example "loopback") and connect to the server.
For version 18.0.1008 to 18.0.1020, use:
http://libjingle.googlecode.com/svn-history/r103/trunk/talk/examples/peerconnection/server/server_test.html
For version 18.0.1021 and later, use:
http://libjingle.googlecode.com/svn/trunk/talk/examples/peerconnection/server/server_test.html
4. Open the test page, connect to the server, select the loopback peer, click
call.
* TWO CLIENTS CALLING EACH OTHER
1. Start peerconnection_server.
2. Start the WebRTC Chrome build, see scenario (1).
3. Open the test page, connect to the server.
4. Open a new new tab, open the test page, connect to the server.
OR
On another machine, repeat 2 and 3.
5. Select the other peer, click call.
* TWO AUTOCALLING CLIENTS
1. Run the procedure for "two clients calling each other", but give the pages
the following parameters:
webrtc_test.html?name=test1&autoconnect=yes&server=localhost:8888
webrtc_test.html?name=test2&autocall=test1&server=localhost:8888

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test/sanity_check/www/html/webrtc_test.html
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<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
<!--
Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Use of this source code is governed by a BSD-style license
that can be found in the LICENSE file in the root of the source
tree. An additional intellectual property rights grant can be found
in the file PATENTS. All contributing project authors may
be found in the AUTHORS file in the root of the source tree.
-->
<html>
<head>
<title>WebRTC Test</title>
<style type="text/css">
body, input, button, select, table {
font-family:"Lucida Grande", "Lucida Sans", Verdana, Arial, sans-serif;
font-size: 13 px;
}
body, input:enable, button:enable, select:enable, table {
color: rgb(51, 51, 51);
}
h1 {font-size: 40 px;}
</style>
<script type="text/javascript">
// TODO(henrikg): Catch more exceptions
var server;
var myId = -1;
var myName;
var remoteId = -1;
var remoteName;
var request = null;
var hangingGet = null;
var pc = null;
var localStream = null;
var disconnecting = false;
var callState = 0; // 0 - Not started, 1 - Call ongoing
var startupAction = "";
var autoCallName = "";
var autoCallTries = 0;
// General
function parseUrlParameters() {
window.location.href.replace(/[?&]+([^=&]+)=([^&]*)/gi,
function(m, key, value) {
// Some specific parameters.
if ((key == "autoconnect") && (value == "yes")) {
startupAction = "connect";
} else if (key == "autocall") {
startupAction = "call";
autoCallName = unescape(value);
// The rest are set to elements with corresponding id.
} else {
document.getElementById(key).value = unescape(value);
}
});
}
function trace(txt) {
var elem = document.getElementById("debug");
elem.innerHTML += txt + "<br>";
}
function trace_warning(txt) {
var wtxt = "<b>" + txt + "</b>";
trace(wtxt);
}
function trace_exception(e, txt) {
var etxt = "<b>" + txt + "</b> (" + e.name + " / " + e.message + ")";
trace(etxt);
}
function setCallState(state) {
trace("Changing call state: " + callState + " -> " + state);
callState = state;
}
function checkPeerConnection() {
if (!pc) {
trace_warning("No PeerConnection object exists");
return 0;
}
return 1;
}
// Local stream generation
function gotStream(s) {
var url = webkitURL.createObjectURL(s);
document.getElementById("localView").src = url;
trace("User has granted access to local media. url = " + url);
localStream = s;
if (startupAction == "connect" || startupAction == "call")
connect();
}
function gotStreamFailed(error) {
alert("Failed to get access to local media. Error code was " + error.code +
".");
trace_warning("Failed to get access to local media. Error code was " +
error.code);
}
function getUserMedia() {
try {
navigator.webkitGetUserMedia("video,audio", gotStream, gotStreamFailed);
trace("Requested access to local media");
} catch (e) {
trace_exception(e, "getUserMedia error");
}
}
// Peer list and remote peer handling
function peerExists(id) {
try {
var peerList = document.getElementById("peers");
for (var i = 0; i < peerList.length; i++) {
if (parseInt(peerList.options[i].value) == id)
return true;
}
} catch (e) {
trace_exception(e, "Error searching for peer");
}
return false;
}
function addPeer(id, pname) {
try {
var peerList = document.getElementById("peers");
var option = document.createElement("option");
option.text = pname;
option.value = id;
peerList.add(option, null);
} catch (e) {
trace_exception(e, "Error adding peer");
}
}
function removePeer(id) {
try {
var peerList = document.getElementById("peers");
for (var i = 0; i < peerList.length; i++) {
if (parseInt(peerList.options[i].value) == id) {
peerList.remove(i);
break;
}
}
} catch (e) {
trace_exception(e, "Error removing peer");
}
}
function clearPeerList() {
var peerList = document.getElementById("peers");
while (peerList.length > 0)
peerList.remove(0);
}
function setSelectedPeer(id) {
try {
var peerList = document.getElementById("peers");
for (var i = 0; i < peerList.length; i++) {
if (parseInt(peerList.options[i].value) == id) {
peerList.options[i].selected = true;
return true;
}
}
} catch (e) {
trace_exception(e, "Error setting selected peer");
}
return false;
}
function getPeerName(id) {
try {
var peerList = document.getElementById("peers");
for (var i = 0; i < peerList.length; i++) {
if (parseInt(peerList.options[i].value) == id) {
return peerList.options[i].text;
}
}
} catch (e) {
trace_exception(e, "Error finding peer name");
return;
}
return;
}
function getPeerId(peer_name) {
try {
var peerList = document.getElementById("peers");
for (var i = 0; i < peerList.length; i++) {
if (peerList.options[i].text == peer_name) {
return parseInt(peerList.options[i].value);
}
}
} catch (e) {
trace_exception(e, "Error finding peer ID");
return -1;
}
return -1;
}
function storeRemoteInfo() {
try {
var peerList = document.getElementById("peers");
if (peerList.selectedIndex < 0) {
alert("Please select a peer.");
return false;
} else
remoteId = parseInt(peerList.options[peerList.selectedIndex].value);
remoteName = peerList.options[peerList.selectedIndex].text;
} catch (e) {
trace_exception(e, "Error storing remote peer info");
return false;
}
return true;
}
// Call control
function createPeerConnection() {
if (pc) {
trace_warning("PeerConnection object already exists");
}
trace("Creating PeerConnection object");
try {
pc = new webkitDeprecatedPeerConnection("STUN stun.l.google.com:19302",
onSignalingMessage);
pc.onaddstream = onAddStream;
pc.onremovestream = onRemoveStream;
} catch (e) {
trace_exception(e, "Create PeerConnection error");
}
}
function doCall() {
if (!storeRemoteInfo())
return;
document.getElementById("call").disabled = true;
document.getElementById("peers").disabled = true;
createPeerConnection();
trace("Adding stream");
pc.addStream(localStream);
document.getElementById("hangup").disabled = false;
setCallState(1);
}
function hangUp() {
document.getElementById("hangup").disabled = true;
trace("Sending BYE to " + remoteName + " (ID " + remoteId + ")");
sendToPeer(remoteId, "BYE");
closeCall();
}
function closeCall() {
trace("Stopping showing remote stream");
document.getElementById("remoteView").src = "dummy";
if (pc) {
trace("Stopping call [pc.close()]");
pc.close();
pc = null;
} else
trace("No pc object to close");
remoteId = -1;
document.getElementById("call").disabled = false;
document.getElementById("peers").disabled = false;
setCallState(0);
}
function autoCall() {
var peer_id = getPeerId(autoCallName);
if (peer_id < 0) {
// Retry a couple of times before giving up.
if (autoCallTries < 3)
window.setTimeout(autoCall, ++autoCallTries * 1000);
else
trace_warning("Could not find a peer with name " + autoCallName +
", giving up");
return;
}
setSelectedPeer(peer_id);
doCall(0);
}
// PeerConnection callbacks
function onAddStream(e) {
var stream = e.stream;
var url = webkitURL.createObjectURL(stream);
document.getElementById("remoteView").src = url;
trace("Started showing remote stream. url = " + url);
}
function onRemoveStream(e) {
// Currently if we get this callback, call has ended.
document.getElementById("remoteView").src = "";
trace("Stopped showing remote stream");
}
function onSignalingMessage(msg) {
trace("Sending message to " + remoteName + " (ID " + remoteId + "):\n" + msg);
sendToPeer(remoteId, msg);
}
// TODO(henrikg): Add callbacks onconnecting, onopen and onstatechange.
// Server interaction
function handleServerNotification(data) {
trace("Server notification: " + data);
var parsed = data.split(",");
if (parseInt(parsed[2]) == 1) { // New peer
var peerId = parseInt(parsed[1]);
if (!peerExists(peerId)) {
var peerList = document.getElementById("peers");
if (peerList.length == 1 && peerList.options[0].value == -1)
clearPeerList();
addPeer(peerId, parsed[0]);
document.getElementById("peers").disabled = false;
document.getElementById("call").disabled = false;
}
} else if (parseInt(parsed[2]) == 0) { // Removed peer
removePeer(parseInt(parsed[1]));
if (document.getElementById("peers").length == 0) {
document.getElementById("peers").disabled = true;
addPeer(-1, "No other peer connected");
}
}
}
function handlePeerMessage(peer_id, msg) {
var peerName = getPeerName(peer_id);
if (peerName == undefined) {
trace_warning("Received message from unknown peer (ID " + peer_id +
"), ignoring message:");
trace(msg);
return;
}
trace("Received message from " + peerName + " (ID " + peer_id + "):\n" + msg);
// Assuming we receive the message from the peer we want to communicate with.
// TODO(henrikg): Only accept messages from peer we communicate with with if
// call is ongoing.
if (msg.search("BYE") == 0) {
// Other side has hung up.
document.getElementById("hangup").disabled = true;
closeCall()
} else {
if (!pc) {
// Other side is calling us, startup
if (!setSelectedPeer(peer_id)) {
trace_warning("Recevied message from unknown peer, ignoring");
return;
}
if (!storeRemoteInfo())
return;
document.getElementById("call").disabled = true;
document.getElementById("peers").disabled = true;
createPeerConnection();
try {
pc.processSignalingMessage(msg);
} catch (e) {
trace_exception(e, "Process signaling message error");
}
trace("Adding stream");
pc.addStream(localStream);
document.getElementById("hangup").disabled = false;
} else {
try {
pc.processSignalingMessage(msg);
} catch (e) {
trace_exception(e, "Process signaling message error");
}
}
}
}
function getIntHeader(r, name) {
var val = r.getResponseHeader(name);
trace("header value: " + val);
return val != null && val.length ? parseInt(val) : -1;
}
function hangingGetCallback() {
try {
if (hangingGet.readyState != 4 || disconnecting)
return;
if (hangingGet.status != 200) {
trace_warning("server error, status: " + hangingGet.status + ", text: " +
hangingGet.statusText);
disconnect();
} else {
var peer_id = getIntHeader(hangingGet, "Pragma");
if (peer_id == myId) {
handleServerNotification(hangingGet.responseText);
} else {
handlePeerMessage(peer_id, hangingGet.responseText);
}
}
if (hangingGet) {
hangingGet.abort();
hangingGet = null;
}
if (myId != -1)
window.setTimeout(startHangingGet, 0);
} catch (e) {
trace_exception(e, "Hanging get error");
}
}
function onHangingGetTimeout() {
trace("hanging get timeout. issuing again");
hangingGet.abort();
hangingGet = null;
if (myId != -1)
window.setTimeout(startHangingGet, 0);
}
function startHangingGet() {
try {
hangingGet = new XMLHttpRequest();
hangingGet.onreadystatechange = hangingGetCallback;
hangingGet.ontimeout = onHangingGetTimeout;
hangingGet.open("GET", server + "/wait?peer_id=" + myId, true);
hangingGet.send();
} catch (e) {
trace_exception(e, "Start hanging get error");
}
}
function sendToPeer(peer_id, data) {
if (myId == -1) {
alert("Not connected.");
return;
}
if (peer_id == myId) {
alert("Can't send a message to oneself.");
return;
}
var r = new XMLHttpRequest();
r.open("POST", server + "/message?peer_id=" + myId + "&to=" + peer_id, false);
r.setRequestHeader("Content-Type", "text/plain");
r.send(data);
r = null;
}
function signInCallback() {
try {
if (request.readyState == 4) {
if (request.status == 200) {
var peers = request.responseText.split("\n");
myId = parseInt(peers[0].split(",")[1]);
trace("My id: " + myId);
clearPeerList();
var added = 0;
for (var i = 1; i < peers.length; ++i) {
if (peers[i].length > 0) {
trace("Peer " + i + ": " + peers[i]);
var parsed = peers[i].split(",");
addPeer(parseInt(parsed[1]), parsed[0]);
++added;
}
}
if (added == 0)
addPeer(-1, "No other peer connected");
else {
document.getElementById("peers").disabled = false;
document.getElementById("call").disabled = false;
}
startHangingGet();
request = null;
document.getElementById("connect").disabled = true;
document.getElementById("disconnect").disabled = false;
if (startupAction == "call") {
startupAction = "";
window.setTimeout(autoCall, 1000);
}
}
}
} catch (e) {
trace_exception(e, "Sign in error");
document.getElementById("connect").disabled = false;
}
}
function signIn() {
try {
request = new XMLHttpRequest();
request.onreadystatechange = signInCallback;
request.open("GET", server + "/sign_in?" + myName, true);
request.send();
} catch (e) {
trace_exception(e, "Start sign in error");
document.getElementById("connect").disabled = false;
}
}
function connect() {
myName = document.getElementById("name").value.toLowerCase();
server = document.getElementById("server").value.toLowerCase();
if (myName.length == 0) {
alert("I need a name please.");
document.getElementById("local").focus();
} else {
// TODO(henrikg): Disable connect button here, but we need a timeout and
// check if we have connected, if so enable it again.
signIn();
}
}
function disconnect() {
if (callState == 1)
hangUp();
disconnecting = true;
if (request) {
request.abort();
request = null;
}
if (hangingGet) {
hangingGet.abort();
hangingGet = null;
}
if (myId != -1) {
request = new XMLHttpRequest();
request.open("GET", server + "/sign_out?peer_id=" + myId, false);
request.send();
request = null;
myId = -1;
}
clearPeerList();
addPeer(-1, "Not connected");
document.getElementById("connect").disabled = false;
document.getElementById("disconnect").disabled = true;
document.getElementById("peers").disabled = true;
document.getElementById("call").disabled = true;
disconnecting = false;
}
// Window event handling
window.onload = function() {
if (navigator.webkitGetUserMedia) {
document.getElementById('testApp').hidden = false;
parseUrlParameters();
getUserMedia();
} else {
document.getElementById('errorText').hidden = false;
}
}
window.onbeforeunload = disconnect;
</script>
</head>
<body>
<h1>WebRTC</h1>
<section id="errorText" hidden="true">
Could not detect WebRTC support.<p>
You must have WebRTC enabled Chrome browser to use this test page. The browser
must be started with the --enable-media-stream command line flag. For more
information, please see
<a href="https://sites.google.com/site/webrtc/blog/webrtcnowavailableinthechromedevchannel">
this blog post</a>.
</section>
<section id="testApp" hidden="true">
<table border="0">
<tr>
<td>Local Preview</td>
<td>Remote Video</td>
</tr>
<tr>
<td>
<video width="320" height="240" id="localView" autoplay="autoplay"></video>
</td>
<td>
<video width="640" height="480" id="remoteView" autoplay="autoplay"></video>
</td>
</tr>
</table>
<table border="0">
<tr>
<td valign="top">
<table border="0" cellpaddning="0" cellspacing="0">
<tr>
<td>Server:</td>
<td>
<input type="text" id="server" size="30" value="http://localhost:8888"/>
</td>
</tr>
<tr>
<td>Name:</td><td><input type="text" id="name" size="30" value="name"/></td>
</tr>
</table>
</td>
<td valign="top">
<button id="connect" onclick="connect();">Connect</button><br>
<button id="disconnect" onclick="disconnect();" disabled="true">Disconnect
</button>
</td>
<td>&nbsp;&nbsp;&nbsp;</td>
<td valign="top">
Connected peers:<br>
<select id="peers" size="5" disabled="true">
<option value="-1">Not connected</option>
</select>
</td>
<td valign="top">
<!--input type="text" id="peer_id" size="3" value="1"/><br-->
<button id="call" onclick="doCall();" disabled="true">Call</button><br>
<button id="hangup" onclick="hangUp();" disabled="true">Hang up</button><br>
</td>
</tr>
</table>
<button onclick="document.getElementById('debug').innerHTML='';">Clear log
</button>
<pre id="debug"></pre>
</section>
</body>
</html>