README

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Getting started

Installing Locomote

Install Locomote using Bower with the following command:

bower install locomote

Install Locomote using npm with the following command:

npm install locomote-video-player

Running Locomote

To run Locomote in a web page, you need to host both the SWF (Player.swf in example below), and the JavaScript library (locomote.min.js in example below). Use a simple page like:

<html>
  <head>
    <title>Locomote</title>
    <style type="text/css">
      #player {
        width: 320px;
        height: 240px;
      }
    </style>
    <script src="locomote.min.js"></script>
    <script src="http://code.jquery.com/jquery-2.1.1.min.js"></script>
    <script type="text/javascript">

      $(document).ready(function() {
        /* Load SWF and instantiate Locomote */
        var locomote = new Locomote('player', 'Player.swf');

        /* Set up a listener for when the API is ready to be used */
        locomote.on('apiReady', function() {
          console.log('API is ready. `play` can now be called');

          /* Tell Locomote to play the specified media */
          locomote.play('rtsp://server.com/stream');
        });

        /* Start listening for streamStarted event */
        locomote.on('streamStarted', function() {
          console.log('stream has started');
        });

        /* If any error occurs, we should take action */
        locomote.on('error', function(err) {
          console.log(err);
        });
      });

    </script>
  </head>
  <body>
    <div id="player" class="player"></div>
  </body>
</html>

Socket Policy Server

Locomote uses sockets to connect to RTSP video streams which requires a socket policy server to be implemented. For RTMP and HTTP streams no socket policy server is required.

Flash Player 9 and above implements a strict access policy for Flash applications that make Socket or XMLSocket connections to a remote host. It now requires the presence of a socket policy file on the server.

When the Flash Player tries to make a connection, it checks in two places for the socket policy:

Port 843. If you are the administrator of a server, you can set up an application to listen on this port and return a server-wide socket policy.
The destination port. If you're running your own xml server, you can configure it to send the socket policy file.

The Flash player always tries port 843 first; if there's no response after 3 seconds, then it tries the destination port.

When the Flash player makes a connection, it sends the following XML string to the server:

<policy-file-request/>

Your server then must send the following XML in reply:

<cross-domain-policy>
     <allow-access-from domain="*" to-ports="*" />
</cross-domain-policy>

* is the wildcard and means "all ports/domains". If you want to restrict access to a particular port, enter the port number, or a list or range of numbers.

For more info about socket policy files and how to set up a server please read the following articles:

Setting up a socket policy file server

Policy file changes in Flash Player 9 and Flash Player 10

API Specification

Construction / Destruction

Locomote(element, url)

Locomote player constructor. Will load the Locomote SWF and embed the Locomote player in a DOM element.

First argument is either an ID to an element in the DOM as a string or a reference to a DOM element. This is where Locomote will embed the player.

The second argument is the URL to the player SWF.

The player will load asynchronously. When the player is loaded an apiReady event is sent. Before the apiReady event, no API methods can be used except on and off.

destroy()

Will remove the tag from the element is was embedded to and remove all references to it held by the javascript library. This can be called as any other action. E.g.

var locomote = new Locomote('player', 'Player.swf');
locomote.destroy();

Actions

play(url:String, [options:Object])

Starts playing video from url. Protocol is determined by url. Example: rtsp://server:port/stream.

Supported protocols:

rtsp - RTSP over TCP

rtsph - RTSP over HTTP

rtsphs - RTSP over HTTPS

rtsphap - RTSP over HTTPS via Axis Proxy

rtmp - RTMP

rtmpt - RTMP over HTTP

rtmps - RTMP over SSL

http - Progressive download via HTTP

https - Progressive download via HTTP over SSL

httpm - MJPEG over HTTP (via multipart/x-mixed-replace)

options is an optional object with the following attributes:

offset - The offset to start the stream at. This is only supported by the rtsp[h|hs|hap] protocol and requires the RTSP server to respect the range header in the play request.

httpUrl - The URL to use in HTTP requests if it differs from the RTSP URL. This is only supported by the rtsp[h|hs] (Note: not supported by rtsphap) protocol

stop()

Stops video stream.

seek(offset)

Seeks to the position specified by offset (calculated from the start of stream).

If the currently player stream is RTMP, it may not work with seeking if the stream is live. Even if the material played is recorded it may not work depending on RTMP server implementation. In the RTMP case, this is really just delegated to the implementation in the NetStream class.

This does not work for RTSP at all (yet).

pause()

Pauses video stream.

resume()

Resumes video from paused state.

playFrames(timestamp)

Appends all received frames up to and including the given timestamp to the play buffer. Only applicable if player is configured with frameByFrame.

streamStatus()

Returns a status object with the following data (if an entry is unknown, that value will be null):

fps - frames per second.

resolution (object) - the stream size { width, height }.

playbackSpeed - current playback speed. 1.0 is normal stream speed.

current time - ms from start of stream.

protocol - which high-level transport protocol is in use.

state - current playback state (playing, paused, stopped).

streamURL - the source of the current media.

duration - the duration of the currently playing media, or -1 if not available

playerStatus()

Returns a status object with the following data:

buffer - The length of the buffer in seconds.

microphoneVolume - the volume of the microphone when capturing audio

speakerVolume - the volume of the speakers (i.e. the stream volume).

microphoneMuted (bool) - if the microphone is muted.

speakerMuted (bool) - if the speakers are muted.

fullScreen (bool) - if the player is currently in fullscreen mode.

version - the Locomote version number.

speakerVolume(vol)

Sets speaker volume from 0-100. The default value is 50.

muteSpeaker()

Mutes the speaker volume. Remembers the current volume and resets to it if the speakers are unmuted.

unmuteSpeaker()

Resets the volume to previous unmuted value.

microphoneVolume(vol)

Sets microphone volume from 0-100. The default value is 50.

muteMicrophone()

Mutes the microphone. Remembers the current volume and resets to it if the microphone is unmuted.

unmuteMicrophone()

Resets the volume to previous unmuted value.

startAudioTransmit(url, type)

Starts transmitting microphone input to the camera speaker. The optional type parameter can be used for future implementations of other protocols, currently only the Axis audio transmit api is supported. For Axis cameras the url parameter should be in the format - http://server:port/axis-cgi/audio/transmit.cgi.

If the user must grant permission to use the microphone an audioTransmitRequestPermission event will be dispatched and startAudioTransmit must be called again once permission has been granted.

stopAudioTransmit()

Stops transmitting microphone input to the camera speaker.

config(config)

Sets configuration values of the player. config is a JavaScript object that can have the following optional values:

buffer - The number of seconds that should be buffered. The default value is 3.

connectionTimeout - The number of seconds before a broken connection times out and is closed. The default value is 10.

keepAlive - The number of seconds between keep alive requests (only RTSP at the moment). The default value is 0 (disabled).

scaleUp - Specifies if the video can be scaled up or not. The default value is false.

allowFullscreen - Specifices if fullscreen mode is allowed or not. The default value is true.

debugLogger - Specifices if debug messages should be shown in the Flash console or not. The default value is false.

frameByFrame - Specifices if media should be played immediately or wait for calls to playFrames. Not supported by the rtmp protocol. The default value is false. The http and https protocol implements this by creating virtual frames, a timestamp given in the frameReady event may not correspond to a real video frame, and the player may play up to 50 ms more than the last playFrames call specified. The rtsp[h|hs|hap] protocol dispatches the frameReady event for each assembled FLV tag, if audio and video is received out of order this will cause frameReady events to be dispatched out of order.

on(eventName:String, callback:Function)

Starts listening for events with eventName. Calls callback when event triggers.

off(eventName:String, callback:Function)

Stops listening for events with eventName.

Events

apiReady

Dispatched when the player is fully initialized. This is always the first event to be sent. Before the apiReady event no API methods can be called except on and off.

streamStarted

Dispatched when video streams starts.

streamPaused(result)

Dispatched when video stream is paused. result is an object with a single property reason that can have the following values:

user - stream was paused by user.

buffering - stream has stopped for buffering.

streamStopped

Dispatched when stream stops.

frameReady(timestamp)

Dispatched when a new frame, or pseudo-frame, is available to be appended to the play buffer. The timestamp of the frame is given by the argument. Append it using the playFrames method. This event will only be dispatched if the player is configured with the frameByFrame option. Otherwise, all frames will be appended to the play buffer immediately when received and this event will not be dispatched.

error(error)

Dispatched when video stream fails. error can be either protocol error (rtsp etc) or Locomote internal error. error is a generic object.

Locomote reports the following types of errors:

RTSP - The default error codes that are sent from the RTSP stream. Error codes: 100 - 551.

Flash Player - Errors that are reported by Flash Socket, NetStream and NetConnection classes. Error codes: 700 - 799.

Locomote - Errors generated by the Locomote player. Error codes: 800 - 899.

For detailed information about the errors, please see the ErrorManager class.

audioTransmitStarted

Dispatched when audio transmission starts.

audioTransmitStopped

Dispatched when audio transmission stops.

audioTransmitRequestPermission

Dispatched when flash is prompting the user to grant or deny access to the microphone. When this event is dispatched the setup is aborted and startAudioTransmit must be called again after the event audioTransmitAllowed has been dispatched.

audioTransmitAllowed

Dispatched when user has granted permission to use the microphone. A new call to startAudioTransmit must be made to initiate audio transmission.

audioTransmitDenied

Dispatched when user has denied permission to use the microphone. If this event is fired, any future calls to startAudioTransmit will generate an error (816).

fullscreenEntered

Dispatched when the player enters fullscreen mode.

fullscreenExited

Dispatched when the player exits fullscreen mode.

log

Dispatched when a log message is sent from the player.

Building Locomote

Building with npm

To compile the project, nodejs and npm is required. Since npm is bundled with nodejs, you only need to download and install nodejs.

To build Locomote, simply run npm install followed by gulp in the root directory. This will download Adobe Flex SDK, and other required modules and build the SWF file and the JavaScript library to dist/.

Building with Flash Builder

It's also possible to build Locomote with Flash Builder. Follow the steps below to set up a Flash Builder project.

Clone the Locomote repository from Github.
Build the project with npm as described above. This will build as3corelib and the VERSION file which are both required dependencies.
Create a new ActionScript project from Flash Builder and save it in the root folder of the cloned repository.
Inside Flash Builder, right click the Player.as file that is now in the default package and select "Set as Default Application".
Remove the .as file with the same name you used for the project that was automatically created in default package.
Add as3corelib to the project by selecting "Properties" in the "Project menu" and then "ActionScript Build Path". Click "Add SWC..." and add as3corelib which is located here: /ext/as3corelib/bin/as3corelib.swc. Make sure that the library is merged into the code. Please note that the as3corelib.swc file will only be available after you have built the project with npm.
You may need to change the path to the default HTML file in "Run/Debug Settings". Edit the Player launch configuration and make sure that the correct url to the HTML file is selected.
The project can now be built by Flash Builder. Please note that you also need to modify the default HTML template provided with the Flash Builder project to load the swf and Javascript file properly. An example of a minimal HTML file is provided below.

The Flash Builder project files and build folders will be ignored by git automatically so you shouldn't have to add anything to the repository after setting up the project.

License

This project is licensed under the BSD 3-Clause License. See LICENSE file.

Usage no npm install needed!