Category Archives: Open Source

Progress with rtc.io

Digital Media, html5, Open Source, rtc.io, standards, WebRTC, , , ,

At the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.

webrtc_talk

Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.

WDCNZ presentation page5

One of the most exciting opportunities is still under-exploited: the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.

Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.

rtcio_modules

We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections:

Packages per day across popular platforms (Shamelessly copied from: http://blog.nodejitsu.com/npm-innovation-through-modularity/)

For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.

For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.

Using rtc.io rtc JS library
Using rtc.io rtc JS library

So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course: enjoy WebRTC! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team!

On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating!!

Use deck.js as a remote presentation tool

code, Digital Media, html5, LCA, Open Source, WebRTC, , , , ,

deck.js is one of the new HTML5-based presentation tools. It’s simple to use, in particular for your basic, every-day presentation needs. You can also create more complex slides with animations etc. if you know your HTML and CSS.

Yesterday at linux.conf.au (LCA), I gave a presentation using deck.js. But I didn’t give it from the lectern in the room in Perth where LCA is being held – instead I gave it from the comfort of my home office at the other end of the country.

I used my laptop with in-built webcam and my Chrome browser to give this presentation. Beforehand, I had uploaded the presentation to a Web server and shared the link with the organiser of my speaker track, who was on site in Perth and had set up his laptop in the same fashion as myself. His screen was projecting the Chrome tab in which my slides were loaded and he had hooked up the audio output of his laptop to the room speaker system. His camera was pointed at the audience so I could see their reaction.

I loaded a slide master URL:
http://html5videoguide.net/presentations/lca_2014_webrtc/?master
and the room loaded the URL without query string:
http://html5videoguide.net/presentations/lca_2014_webrtc/.

Then I gave my talk exactly as I would if I was in the same room. Yes, it felt exactly as though I was there, including nervousness and audience feedback.

How did we do that? WebRTC (Web Real-time Communication) to the rescue, of course!

We used one of the modules of the rtc.io project called rtc-glue to add the video conferencing functionality and the slide navigation to deck.js. It was actually really really simple!

Here are the few things we added to deck.js to make it work:

  • Code added to index.html to make the video connection work: 
    <meta name="rtc-signalhost" content="http://rtc.io/switchboard/">
<meta name="rtc-room" content="lca2014">
...
<video id="localV" rtc-capture="camera" muted></video>
<video id="peerV" rtc-peer rtc-stream="localV"></video>
...
<script src="glue.js"></script>
<script>
glue.config.iceServers = [{ url: 'stun:stun.l.google.com:19302' }];
</script>

    The iceServers config is required to punch through firewalls – you may also need a TURN server. Note that you need a signalling server – in our case we used http://rtc.io/switchboard/, which runs the code from rtc-switchboard.

  • Added glue.js library to deck.js:

    Downloaded from https://raw.github.com/rtc-io/rtc-glue/master/dist/glue.js into the source directory of deck.js.

  • Code added to index.html to synchronize slide navigation: 
    glue.events.once('connected', function(signaller) {
  if (location.search.slice(1) !== '') {
    $(document).bind('deck.change', function(evt, from, to) {
      signaller.send('/slide', {
        idx: to,
        sender: signaller.id
      });
    });
  }
  signaller.on('slide', function(data) {
    console.log('received notification to change to slide: ', data.idx);
    $.deck('go', data.idx);
  });
});

    This simply registers a callback on the slide master end to send a slide position message to the room end, and a callback on the room end that initiates the slide navigation.

And that’s it!

You can find my slide deck on GitHub.

Feel free to write your own slides in this manner – I would love to have more users of this approach. It should also be fairly simple to extend this to share pointer positions, so you can actually use the mouse pointer to point to things on your slides remotely. Would love to hear your experiences!

Note that the slides are actually a talk about the rtc.io project, so if you want to find out more about these modules and what other things you can do, read the slide deck or watch the talk when it has been published by LCA.

Many thanks to Damon Oehlman for his help in getting this working.

BTW: somebody should really fix that print style sheet for deck.js – I’m only ever getting the one slide that is currently showing. 😉

Video Conferencing in HTML5: WebRTC via Socket.io

code, Digital Media, open codecs, Open Source, standards, Videos, WebRTC, , ,

Six months ago I experimented with Web sockets for WebRTC and the early implementations of PeerConnection in Chrome. Last week I gave a presentation about WebRTC at Linux.conf.au, so it was time to update that codebase.

I decided to use socket.io for the signalling following the idea of Luc, which made the server code even smaller and reduced it to a mere reflector:

 var app = require('http').createServer().listen(1337);
 var io = require('socket.io').listen(app);

 io.sockets.on('connection', function(socket) {
         socket.on('message', function(message) {
         socket.broadcast.emit('message', message);
     });
 });

Then I turned to the client code. I was surprised to see the massive changes that PeerConnection has gone through. Check out my slide deck to see the different components that are now necessary to create a PeerConnection.

I was particularly surprised to see the SDP object now fully exposed to JavaScript and thus the ability to manipulate it directly rather than through some API. This allows Web developers to manipulate the type of session that they are asking the browsers to set up. I can imaging e.g. if they have support for a video codec in JavaScript that the browser does not provide built-in, they can add that codec to the set of choices to be offered to the peer. While it is flexible, I am concerned if this might create more problems than it solves. I guess we’ll have to wait and see.

I was also surprised by the need to use ICE, even though in my experiment I got away with an empty list of ICE servers – the ICE messages just got exchanged through the socket.io server. I am not sure whether this is a bug, but I was very happy about it because it meant I could run the whole demo on a completely separate network from the Internet.

The most exciting news since my talk is that Mozilla and Google have managed to get a PeerConnection working between Firefox and Chrome – this is the first cross-browser video conference call without a plugin! The code differences are minor.

Since the specification of the WebRTC API and of the MediaStream API are now official Working Drafts at the W3C, I expect other browsers will follow. I am also looking forward to the possibilities of:

The best places to learn about the latest possibilities of WebRTC are webrtc.org and the W3C WebRTC WG. code.google.com has open source code that continues to be updated to the latest released and interoperable features in browsers.

The video of my talk is in the process of being published. There is a MP4 version on the Linux Australia mirror server, but I expect it will be published properly soon. I will update the blog post when that happens.

Video Conferencing in HTML5: WebRTC via Web Sockets

code, Digital Media, open codecs, Open Source, standards, Videos, ,

A bit over a week ago I gave a presentation at Web Directions Code 2012 in Melbourne. Maxine and John asked me to speak about something related to HTML5 video, so I went for the new shiny: WebRTC – real-time communication in the browser.

Presentation slides

I only had 20 min, so I had to make it tight. I wanted to show off video conferencing without special plugins in Google Chrome in just a few lines of code, as is the promise of WebRTC. To a large extent, I achieved this. But I made some interesting discoveries along the way. Demos are in the slide deck.

UPDATE: Opera 12 has been released with WebRTC support.

Housekeeping: if you want to replicate what I have done, you need to install a Google Chrome Web Browser 19+. Then make sure you go to chrome://flags and activate the MediaStream and PeerConnection experiment(s). Restart your browser and now you can experiment with this feature. Big warning up-front: it’s not production-ready, since there are still changes happening to the spec and there is no compatible implementation by another browser yet.

Here is a brief summary of the steps involved to set up video conferencing in your browser:

  1. Set up a video element each for the local and the remote video stream.
  2. Grab the local camera and stream it to the first video element.
  3. (*) Establish a connection to another person running the same Web page.
  4. Send the local camera stream on that peer connection.
  5. Accept the remote camera stream into the second video element.

Now, the most difficult part of all of this – believe it or not – is the signalling part that is required to build the peer connection (marked with (*)). Initially I wanted to run completely without a server and just enter the remote’s IP address to establish the connection. This is, however, not a functionality that the PeerConnection object provides [might this be something to add to the spec?].

So, you need a server known to both parties that can provide for the handshake to set up the connection. All the examples that I have seen, such as https://apprtc.appspot.com/, use a channel management server on Google’s appengine. I wanted it all working with HTML5 technology, so I decided to use a Web Socket server instead.

I implemented my Web Socket server using node.js (code of websocket server). The video conferencing demo is in the slide deck in an iframe – you can also use the stand-alone html page. Works like a treat.

While it is still using Google’s STUN server to get through NAT, the messaging for setting up the connection is running completely through the Web Socket server. The messages that get exchanged are plain SDP message packets with a session ID. There are OFFER, ANSWER, and OK packets exchanged for each streaming direction. You can see some of it in the below image:

WebRTC demo

I’m not running a public WebSocket server, so you won’t be able to see this part of the presentation working. But the local loopback video should work.

At the conference, it all went without a hitch (while the wireless played along). I believe you have to host the WebSocket server on the same machine as the Web page, otherwise it won’t work for security reasons.

A whole new world of opportunities lies out there when we get the ability to set up video conferencing on every Web page – scary and exciting at the same time!

My crazy linux.conf.au week

Digital Media, LCA, open codecs, Open Source, standards, video accessibility, , , , , ,

In January I attended the annual Australian Linux and Open Source conference (LCA). But since I was sick all of January and had a lot to catch up on, I never got around to sharing all the talks that I gave during that time.

Drupal Down Under

It started with a talk at Drupal Down Under, which happened the weekend before LCA. I gave a talk titled “HTML5 video specifications” (video, slides).

I spoke about the video and audio element in HTML5, how to provide fallback content, how to encode content, how to control them from JavaScript, and briefly about Drupal video modules, though the next presentation provided much more insight into those. I explained how to make the HTML5 media elements accessible, including accessible controls, captions, audio descriptions, and the new WebVTT file format. I ran out of time to introduce the last section of my slides which are on WebRTC.

Linux.conf.au

On the first day of LCA I gave a talk both in the Multimedia Miniconf and the Browser Miniconf.

Browser Miniconf

In the Browser Miniconf I talked about “Web Standardisation – how browser vendors collaborate, or not” (slides). Maybe the most interesting part about this was that I tried out a new slide “deck” tool called impress.js. I’m not yet sure if I like it but it worked well for this talk, in which I explained how the HTML5 spec is authored and who has input.

I also sat on a panel of browser developers in the Browser Miniconf (more as a standards than as a browser developer, but that’s close enough). We were asked about all kinds of latest developments in HTML5, CSS3, and media standards in the browser.

Multimedia Miniconf

In the Multimedia Miniconf I gave a “HTML5 media accessibility update” (slides). I talked about the accessibility problems of Flash, how native HTML5 video players will be better, about accessible video controls, captions, navigation chapters, audio descriptions, and WebVTT. I also provided a demo of how to synchronize multiple video elements using a polyfill for the multitrack API.

I also provided an update on HTTP adaptive streaming APIs as a lightning talk in the Multimedia Miniconf. I used an extract of the Drupal conference slides for it.

Main conference

Finally, and most importantly, Alice Boxhall and myself gave a talk in the main linux.conf.au titled “Developing Accessible Web Apps – how hard can it be?” (video, slides). I spoke about a process that you can follow to make your Web applications accessible. I’m writing a separate blog post to explain this in more detail. In her part, Alice dug below the surface of browsers to explain how the accessibility markup that Web developers provide is transformed into data structures that are handed to accessibility technologies.

Open Media Developers Track at OVC 2011

Digital Media, FOMS, open codecs, Open Source, standards, video accessibility, , , , , , , , , ,

The Open Video Conference that took place on 10-12 September was so overwhelming, I’ve still not been able to catch my breath! It was a dense three days for me, even though I only focused on the technology sessions of the conference and utterly missed out on all the policy and content discussions.

Roughly 60 people participated in the Open Media Software (OMS) developers track. This was an amazing group of people capable and willing to shape the future of video technology on the Web:

  • HTML5 video developers from Apple, Google, Opera, and Mozilla (though we missed the NZ folks),
  • codec developers from WebM, Xiph, and MPEG,
  • Web video developers from YouTube, JWPlayer, Kaltura, VideoJS, PopcornJS, etc.,
  • content publishers from Wikipedia, Internet Archive, YouTube, Netflix, etc.,
  • open source tool developers from FFmpeg, gstreamer, flumotion, VideoLAN, PiTiVi, etc,
  • and many more.

To provide a summary of all the discussions would be impossible, so I just want to share the key take-aways that I had from the main sessions.

WebRTC: Realtime Communications and HTML5

Tim Terriberry (Mozilla), Serge Lachapelle (Google) and Ethan Hugg (CISCO) moderated this session together (slides). There are activities both at the W3C and at IETF – the ones at IETF are supposed to focus on protocols, while the W3C ones on HTML5 extensions.

The current proposal of a PeerConnection API has been implemented in WebKit/Chrome as open source. It is expected that Firefox will have an add-on by Q1 next year. It enables video conferencing, including media capture, media encoding, signal processing (echo cancellation etc), secure transmission, and a data stream exchange.

Current discussions are around the signalling protocol and whether SIP needs to be required by the standard. Further, the codec question is under discussion with a question whether to mandate VP8 and Opus, since transcoding gateways are not desirable. Another question is how to measure the quality of the connection and how to report errors so as to allow adaptation.

What always amazes me around RTC is the sheer number of specialised protocols that seem to be required to implement this. WebRTC does not disappoint: in fact, the question was asked whether there could be a lighter alternative than to re-use dozens of years of protocol development – is it over-engineered? Can desktop players connect to a WebRTC session?

We are already in a second or third revision of this part of the HTML5 specification and yet it seems the requirements are still being collected. I’m quietly confident that everything is done to make the lives of the Web developer easier, but it sure looks like a huge task.

The Missing Link: Flash to HTML5

Zohar Babin (Kaltura) and myself moderated this session and I must admit that this session was the biggest eye-opener for me amongst all the sessions. There was a large number of Flash developers present in the room and that was great, because sometimes we just don’t listen enough to lessons learnt in the past.

This session gave me one of those aha-moments: it the form of the Flash appendBytes() API function.

The appendBytes() function allows a Flash developer to take a byteArray out of a connected video resource and do something with it – such as feed it to a video for display. When I heard that Web developers want that functionality for JavaScript and the video element, too, I instinctively rejected the idea wondering why on earth would a Web developer want to touch encoded video bytes – why not leave that to the browser.

But as it turns out, this is actually a really powerful enabler of functionality. For example, you can use it to:

  • display mid-roll video ads as part of the same video element,
  • sequence playlists of videos into the same video element,
  • implement DVR functionality (high-speed seeking),
  • do mash-ups,
  • do video editing,
  • adaptive streaming.

This totally blew my mind and I am now completely supportive of having such a function in HTML5. Together with media fragment URIs you could even leave all the header download management for resources to the Web browser and just request time ranges from a video through an appendBytes() function. This would be easier on the Web developer than having to deal with byte ranges and making sure that appropriate decoding pipelines are set up.

Standards for Video Accessibility

Philip Jagenstedt (Opera) and myself moderated this session. We focused on the HTML5 track element and the WebVTT file format. Many issues were identified that will still require work.

One particular topic was to find a standard means of rendering the UI for caption, subtitle, und description selection. For example, what icons should be used to indicate that subtitles or captions are available. While this is not part of the HTML5 specification, it’s still important to get this right across browsers since otherwise users will get confused with diverging interfaces.

Chaptering was discussed and a particular need to allow URLs to directly point at chapters was expressed. I suggested the use of named Media Fragment URLs.

The use of WebVTT for descriptions for the blind was also discussed. A suggestion was made to use the voice tag <v> to allow for “styling” (i.e. selection) of the screen reader voice.

Finally, multitrack audio or video resources were also discussed and the @mediagroup attribute was explained. A question about how to identify the language used in different alternative dubs was asked. This is an issue because @srclang is not on audio or video, only on text, so it’s a missing feature for the multitrack API.

Beyond this session, there was also a breakout session on WebVTT and the track element. As a consequence, a number of bugs were registered in the W3C bug tracker.

WebM: Testing, Metrics and New features

This session was moderated by John Luther and John Koleszar, both of the WebM Project. They started off with a presentation on current work on WebM, which includes quality testing and improvements, and encoder speed improvement. Then they moved on to questions about how to involve the community more.

The community criticised that communication of what is happening around WebM is very scarce. More sharing of information was requested, including a move to using open Google+ hangouts instead of Google internal video conferences. More use of the public bug tracker can also help include the community better.

Another pain point of the community was that code is introduced and removed without much feedback. It was requested to introduce a peer review process. Also it was requested that example code snippets are published when new features are announced so others can replicate the claims.

This all indicates to me that the WebM project is increasingly more open, but that there is still a lot to learn.

Standards for HTTP Adaptive Streaming

This session was moderated by Frank Galligan and Aaron Colwell (Google), and Mark Watson (Netflix).

Mark started off by giving us an introduction to MPEG DASH, the MPEG file format for HTTP adaptive streaming. MPEG has just finalized the format and he was able to show us some examples. DASH is XML-based and thus rather verbose. It is covering all eventualities of what parameters could be switched during transmissions, which makes it very broad. These include trick modes e.g. for fast forwarding, 3D, multi-view and multitrack content.

MPEG have defined profiles – one for live streaming which requires chunking of the files on the server, and one for on-demand which requires keyframe alignment of the files. There are clear specifications for how to do these with MPEG. Such profiles would need to be created for WebM and Ogg Theora, too, to make DASH universally applicable.

Further, the Web case needs a more restrictive adaptation approach, since the video element’s API is already accounting for some of the features that DASH provides for desktop applications. So, a Web-specific profile of DASH would be required.

Then Aaron introduced us to the MediaSource API and in particular the webkitSourceAppend() extension that he has been experimenting with. It is essentially an implementation of the appendBytes() function of Flash, which the Web developers had been asking for just a few sessions earlier. This was likely the biggest announcement of OVC, alas a quiet and technically-focused one.

Aaron explained that he had been trying to find a way to implement HTTP adaptive streaming into WebKit in a way in which it could be standardised. While doing so, he also came across other requirements around such chunked video handling, in particular around dynamic ad insertion, live streaming, DVR functionality (fast forward), constraint video editing, and mashups. While trying to sort out all these requirements, it became clear that it would be very difficult to implement strategies for stream switching, buffering and delivery of video chunks into the browser when so many different and likely contradictory requirements exist. Also, once an approach is implemented and specified for the browser, it becomes very difficult to innovate on it.

Instead, the easiest way to solve it right now and learn about what would be necessary to implement into the browser would be to actually allow Web developers to queue up a chunk of encoded video into a video element for decoding and display. Thus, the webkitSourceAppend() function was born (specification).

The proposed extension to the HTMLMediaElement is as follows:

partial interface HTMLMediaElement {
  // URL passed to src attribute to enable the media source logic.
  readonly attribute [URL] DOMString webkitMediaSourceURL;

  bool webkitSourceAppend(in Uint8Array data);

  // end of stream status codes.
  const unsigned short EOS_NO_ERROR = 0;
  const unsigned short EOS_NETWORK_ERR = 1;
  const unsigned short EOS_DECODE_ERR = 2;

  void webkitSourceEndOfStream(in unsigned short status);

  // states
  const unsigned short SOURCE_CLOSED = 0;
  const unsigned short SOURCE_OPEN = 1;
  const unsigned short SOURCE_ENDED = 2;

  readonly attribute unsigned short webkitSourceState;
};

The code is already checked into WebKit, but commented out behind a command-line compiler flag.

Frank then stepped forward to show how webkitSourceAppend() can be used to implement HTTP adaptive streaming. His example uses WebM – there are no examples with MPEG or Ogg yet.

The chunks that Frank’s demo used were 150 video frames long (6.25s) and 5s long audio. Stream switching only switched video, since audio data is much lower bandwidth and more important to retain at high quality. Switching was done on multiplexed files.

Every chunk requires an XHR range request – this could be optimised if the connections were kept open per adaptation. Seeking works, too, but since decoding requires download of a whole chunk, seeking latency is determined by the time it takes to download and decode that chunk.

Similar to DASH, when using this approach for live streaming, the server has to produce one file per chunk, since byte range requests are not possible on a continuously growing file.

Frank did not use DASH as the manifest format for his HTTP adaptive streaming demo, but instead used a hacked-up custom XML format. It would be possible to use JSON or any other format, too.

After this session, I was actually completely blown away by the possibilities that such a simple API extension allows. If I wasn’t sold on the idea of a appendBytes() function in the earlier session, this one completely changed my mind. While I still believe we need to standardise a HTTP adaptive streaming file format that all browsers will support for all codecs, and I still believe that a native implementation for support of such a file format is necessary, I also believe that this approach of webkitSourceAppend() is what HTML needs – and maybe it needs it faster than native HTTP adaptive streaming support.

Standards for Browser Video Playback Metrics

This session was moderated by Zachary Ozer and Pablo Schklowsky (JWPlayer). Their motivation for the topic was, in fact, also HTTP adaptive streaming. Once you leave the decisions about when to do stream switching to JavaScript (through a function such a wekitSourceAppend()), you have to expose stream metrics to the JS developer so they can make informed decisions. The other use cases is, of course, monitoring of the quality of video delivery for reporting to the provider, who may then decide to change their delivery environment.

The discussion found that we really care about metrics on three different levels:

  • measuring the network performance (bandwidth)
  • measuring the decoding pipeline performance
  • measuring the display quality

In the end, it seemed that work previously done by Steve Lacey on a proposal for video metrics was generally acceptable, except for the playbackJitter metric, which may be too aggregate to mean much.

Device Inputs / A/V in the Browser

I didn’t actually attend this session held by Anant Narayanan (Mozilla), but from what I heard, the discussion focused on how to manage permission of access to video camera, microphone and screen, e.g. when multiple applications (tabs) want access or when the same site wants access in a different session. This may apply to real-time communication with screen sharing, but also to photo sharing, video upload, or canvas access to devices e.g. for time lapse photography.

Open Video Editors

This was another session that I wasn’t able to attend, but I believe the creation of good open source video editing software and similar video creation software is really crucial to giving video a broader user appeal.

Jeff Fortin (PiTiVi) moderated this session and I was fascinated to later see his analysis of the lifecycle of open source video editors. It is shocking to see how many people/projects have tried to create an open source video editor and how many have stopped their project. It is likely that the creation of a video editor is such a complex challenge that it requires a larger and more committed open source project – single people will just run out of steam too quickly. This may be comparable to the creation of a Web browser (see the size of the Mozilla project) or a text processing system (see the size of the OpenOffice project).

Jeff also mentioned the need to create open video editor standards around playlist file formats etc. Possibly the Open Video Alliance could help. In any case, something has to be done in this space – maybe this would be a good topic to focus next year’s OVC on?

Monday’s Breakout Groups

The conference ended officially on Sunday night, but we had a third day of discussions / hackday at the wonderful New York Lawschool venue. We had collected issues of interest during the two previous days and organised the breakout groups on the morning (Schedule).

In the Content Protection/DRM session, Mark Watson from Netflix explained how their API works and that they believe that all we need in browsers is a secure way to exchange keys and an indicator of protection scheme is used – the actual protection scheme would not be implemented by the browser, but be provided by the underlying system (media framework/operating system). I think that until somebody actually implements something in a browser fork and shows how this can be done, we won’t have much progress. In my understanding, we may also need to disable part of the video API for encrypted content, because otherwise you can always e.g. grab frames from the video element into canvas and save them from there.

In the Playlists and Gapless Playback session, there was massive brainstorming about what new cool things can be done with the video element in browsers if playback between snippets can be made seamless. Further discussions were about a standard playlist file formats (such as XSPF, MRSS or M3U), media fragment URIs in playlists for mashups, and the need to expose track metadata for HTML5 media elements.

What more can I say? It was an amazing three days and the complexity of problems that we’re dealing with is a tribute to how far HTML5 and open video has already come and exciting news for the kind of applications that will be possible (both professional and community) once we’ve solved the problems of today. It will be exciting to see what progress we will have made by next year’s conference.

Thanks go to Google for sponsoring my trip to OVC.

UPDATE: We actually have a mailing list for open media developers who are interested in these and similar topics – do join at http://lists.annodex.net/cgi-bin/mailman/listinfo/foms.

The new FOMS: Open Media Developers at OVC

Digital Media, FOMS, open codecs, Open Source, standards, video accessibility, , ,

Since 2007 I have organised the annual Foundations of Open Media Software (FOMS) developers workshop. Last year it was held for the first time in the northern hemisphere, in fact on the two days straight after the Open Video Conference (OVC).

This year I’m really excited to announce that the workshop will be an integral part of the Open Video Conference on 10-12 September 2011.

FOMS 2011 will take place as the Open Media Developers track at OVC and I would like to see as many if not more open media software developers attend as we had in last year’s FOMS.

Why should you go?

Well, firstly of course the people. As in previous years, we will have some of the key developers in open media software attend – not as celebrities, but to work with other key developers on hard problems and to make progress.

Then, secondly we believe we have some awesome sessions in preparation:

How we run it

I’m actually not quite satisfied with just these sessions. I’d like to be more flexible on how we make the three days a success for everyone. And this implies that there will continue to be room to add more sessions, even while at the conference, and create breakout groups to address really hard issues all the way through the conference.

I insist on this flexibility because I have seen in past years that the most productive outcomes are created by two or three people breaking away from the group, going into a corner and hacking up some demos or solutions to hard problems and taking that momentum away after the workshop.

To allow this to happen, we will have a plenary on the first day during which we will identify who is actually present at the workshop, what they are working on, what sessions they are planning on a attending, and what other topics they are keen to learn about during the conference that may not yet be addressed by existing sessions.

We’ll repeat this exercise on the Monday after all the rest of the conference is finished and we get a quieter day to just focus on being productive.

But is it worth the effort?

As in the past years, whether the workshop is a success for you depends on you and you alone. You have the power to direct what sessions and breakout groups are being created, and you have the possibility to find others at the workshop that share an interest and drag them away for some productive brainstorming or coding.

I’m going to make sure we have an adequate number of rooms available to actually achieve such an environment. I am very happy to have the support of OVC for this and I am assured we have the best location with plenty of space.

Trip sponsorships

As in previous FOMSes, we have again made sure that travel and conference sponsorship is available to community software developers that would otherwise not be able to attend FOMS. We have several such sponsorships and I encourage you to email the FOMS committee or OVC about it. Mention what you’re working on and what you’re interested to take away from OVC and we can give you free entry, hotel and flight sponsorship.

Oh, and don’t forget to Register for OVC!

WordPress plugin for external videos updated

Digital Media, External Videos, Open Source, Videos, , , , ,

Over the last weeks I’ve updated my “external videos” wordpress plugin. I’ve fixed bugs and added some new functionality.

List of changes:

  • fixed a bug in attaching blog posts to videos for link-through from gallery overlays
  • allow re-attaching a different blog post to a video
  • added a shortcode that allows to link straight through to video pages instead of the overlay
  • fixed a bug on retrieval of keyframe for dotsub
  • added option to add the video posts to the site’s RSS feed
  • fixed a bug on image paths for the thickbox
  • made sure whenever a user goes to the admin page that the cron hook is active
  • changed some class names to avoid clashes with other plugins that people reported
  • turned simple_html_dom code into a class of its own to avoid clashes with other plugins that use this code, too
  • cleaning up entered data from surplus white space
  • styling fixes to the overlay on gallery
  • shielding against a bug with no videos on channels to retrieve yet

Download the new plugin version 0.13

Note: there is something weird going on with the wordpress plugins site, which still shows version 0.7 as the current one, but when you download it, it gets the latest version 0.12. If somebody knows how to fix this, that would be awesome. I think it also stops people from auto-updating this plugin, which is sad with this many improvements.
(I think I fixed it by actually changing the version number in the external-videos.php file – how silly of me – and thanks to the WordPress Forum person who pointed it out to me! Download 0.13 now.)

HTML5 Video Presentations at LCA 2011

Digital Media, LCA, open codecs, Open Source, standards, video accessibility, , , , , , ,

Working in the WHAT WG and the W3C HTML WG, you sometimes forget that all the things that are being discussed so heatedly for standardization are actually leading to some really exciting new technologies that not many outside have really taken note of yet.

This week, during the Australian Linux Conference in Brisbane, I’ve been extremely lucky to be able to show off some awesome new features that browser vendors have implemented for the audio and video elements. The feedback that I got from people was uniformly plain surprise – nobody expected browser to have all these capabilities.

The examples that I showed off have mostly been the result of working on a book for almost 9 months of the past year and writing lots of examples of what can be achieved with existing implementations and specifications. They have been inspired by diverse demos that people made in the last years, so the book is linking to many more and many more amazing demos.

Incidentally, I promised to give a copy of the book away to the person with the best idea for a new Web application using HTML5 media. Since we ran out of time, please shoot me an email or a tweet (@silviapfeiffer) within the next 4 weeks and I will send another copy to the person with the best idea. The copy that I brought along was given to a student who wanted to use HTML5 video to display on surfaces of 3D moving objects.

So, let’s get to the talks.

On Monday, I gave a presentation on “Audio and Video processing in HTML5“, which had a strong focus on the Mozilla Audio API.

I further gave a brief lightning talk about “HTML5 Media Accessibility Update“. I am expecting lots to happen on this topic during this year.

Finally, I gave a presentation today on “The Latest and Coolest in HTML5 Media” with a strong focus on video, but also touching on audio and media accessibility.

The talks were streamed live – congrats to Ryan Verner for getting this working with support from Ben Hutchings from DebConf and the rest of the video team. The videos will apparently be available from http://linuxconfau.blip.tv/ in the near future.

UPDATE 4th Feb 2011: And here is my LCA talk …

with subtitles on YouTube:

Your metadata is not my metadata

Digital Media, open codecs, Open Source, standards, video accessibility, Videos, , , ,

Over the last two days we had the Open Subtitles Summit here in New York. It was very exciting to feel the energy in the room to make a change to media accessibility – I am sure we will see much development over the next 12 months. We spoke much about HTML5 video and standards and had many discussions about subtitles, captions, and other accessibility information.

On Wednesday we had a discussion about metadata and I quickly realized that “your metadata is not my metadata”: everyone used the word for something different. So, I suggested to have a metadata discussion on Thursday where we would put a structure onto all of this, identify what kinds of metadata we have and whether and how it should be supported in HTML5 standards.

Our basic findings are very simple and widely accepted. There are three fundamentally different types of metadata:

  • Technical metadata about video: information about the format of the resource – things that can be determined automatically and are non-controversial, such as the width, height, framerate, audio sample rate etc. This information can be used to, e.g. decide if a video is appropriate for a certain device.
  • Semantic metadata about video: semantic information about the video resource – e.g. license, author, publication date, version, attribution, title, description. This information is good for search and identification.
  • Timed semantic metadata: semantic information that is associated with time intervals of the video, not with the full video – e.g. active speaker, location, date-time, objects.

As we talked about this further, however, we identified subclasses of these generic types that are very important to identify because they will be handled differently.

We found that semantic metadata can be separated into universal metadata and domain-specific metadata. Universal metadata is semantic metadata that can basically be applied to any content. There is very little of that and the W3C Media Annotations WG has done a pretty good job in identifying it. Domain-specific metadata is such metadata that only applies to some content, e.g. all the videos about sports have metadata such as game scores, players, or type of sport.

As for adding such metadata into media resources, we discussed that it makes sense to have the universal metadata explicitly spelled out and to have a generic means to associate name-value pairs with resource. Of course it will all be stored in databases, but there was also a requirement to have it encoded into the media resource – and in our discussion case: into external captions or subtitle files.

As for timed metadata – it is possible to separate this into metadata that is only relevant as part of a subtitle or caption file, because the metadata relates to a certain word or a word sequence, and into independent timed metadata that can be stored in, e.g. JSON or some similar format.

Since we are particularly interested in subtitles and captions, the timed metadata that is associated with words or word sequences is particularly important. The most natural metadata that is useful as part of subtitles is of course speaker segmentation. We also identified that hyperlinks to related content are just as important, since it can enable applications such as popcorn.js.

Potentially there is a use for metadata association with any sequence of words in a caption or subtitle, which could be satisfied with the use of a generic markup element for a sequence of words, such that microdata or RDFa may get associated. A request for such a generic means of associating metadata was made. However, the need for it still has to be confirmed with good use cases – the breakout group was out of time as we came to this point. So, leave your ideas for use cases in the requirements – they will help shape standards.