AppRTC : Google’s WebRTC test app and its parameters

If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC: AppRTC.

When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

Here are my favourite parameters:

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

For example, here’s how a stereo, HD loopback test would look like: https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery:

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

Have fun playing with the main and always up-to-date WebRTC application: AppRTC.

UPDATE 12 May 2014

AppRTC now also supports the following bitrate controls:

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

Example usage: https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

Use deck.js as a remote presentation tool

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. ;-)

WebRTC books – a brief review

I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

Rob’s focus is very much on the features required in a typical Web application:

  • video calls
  • audio calls
  • text chats
  • file sharing

In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

Rob then also extends his introduction to WebRTC to two key application areas: education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.

WebVTT Discussions at FOMS

At the recent FOMS (Foundations of Open Media Software and Standards) Developer Workshop, we had a massive focus on WebVTT and the state of its feature set. You will find links to summaries of the individual discussions in the FOMS Schedule page. Here are some of the key results I went away with.

1. WebVTT Regions

The key driving force for improvements to WebVTT continues to be the accurate representation of CEA608/708 captioning. As part of that drive, we’ve introduced regions (the CEA708 “window” concept) to WebVTT. WebVTT regions satisfy multiple requirements of CEA608/708 captions:

  1. support for rollup captions
  2. support for background color and border color on a group of cues independent of the background color of the individual cue
  3. possibility to move a group of cues from one location on screen to a different
  4. support to specify an anchor point and a growth direction for cues when their text size changes
  5. support for specifying a fixed number of lines to be rendered
  6. possibility to specify which region is rendered in front of which other one when regions overlap

While WebVTT regions enable us to satisfy all of the above points, the specification isn’t actually complete yet and some of the above needs aren’t satisfied yet.

We have an open bug to move a region elsewhere. A first discussion at FOMS seemed to to indicate that we’ll have to add syntax for updating a region at a particular time and thus give region definitions a way to be valid only for a certain time frame. I can imagine that the region definitions that we have in the header of the WebVTT file now would have an implicitly defined time frame from the start to the end of the file, but can be overruled by a re-definition anywhere within the WebVTT file. That redefinition needs to provide a start and end time.

We registered a bug to add specifying the width and height of regions (and possibly of cues) by em (i.e. by multiples of the largest character in a font). This should allow us to have the region grow/shrink around the region anchor point with a change of font size by script or a user. em specifications should also be applied to cues – that matches the column count of CEA708/608 better.

When regions overlap, the original region extension spec already suggested a “layer” cue setting. It will be easy to add it.

Another change that we will ultimately need is the “scroll” setting: we will need to introduce support for scrolling text down or from left-to-right or right-to-left, e.g. vertical scrolling text seems to be used in some Chinese caption use cases.

2. Unify Rendering Approach

The introduction of regions created a second code path in the rendering spec with some duplication. At FOMS we discussed if it was possible to unify that. The suggestion is to render all cues into a region. Those that are not part of a region would be rendered into an anonymous region that covers the complete viewport. There may be some consequences to this, e.g. cue settings should be usable across all cues, no matter whether or not part of a region, and avoiding cue overlap may need to be done within regions.

Here’s a rough outline of the path of the new rendering algorithm:

(1) Render the regions:

Specified Region Anonymous Region
Render values as given: Render following values:
  • width
  • lines
  • regionanchor
  • viewportanchor
  • scroll
  • 100%
  • videoheight/lineheight
  • 0,0
  • 0,0
  • none

(2) Render the cues:

  • Create a cue box and put it in its region (anonymous if none given).
  • Calculate position & size of cue box from cue settings (position, line, size).
  • Calculate position of cue text inside cue box from remaining cue settings (vertical, align).

3. Vertical Features

WebVTT includes vertical rendering, both right-to-left and left-to-right. However, regions are not defined for vertical. Eventually, we’re going to have to look at the vertical features of WebVTT with more details and figure out whether the spec is working for them and what real-world requirements we have missed. We hope we can get some help from users in countries where vertically rendered captions/subtitles are the norm.

4. Best Practices

Some of he WebVTT users at FOMS suggested it would be advantageous to start a list of “best practices” for how to author captions with WebVTT. Example recommendations are:

  • Use line numbers only to position cues from top or bottom of viewport. Don’t use otherwise.
  • Note that when the user increases the fontsize in rollup captions and thus introduces new line breaks, your cues will roll by faster because the number of lines of a rollup is fixed.
  • Make sure to use &lrm; and &rlm; UTF-8 markers to control the directionality of your text.

It would be nice if somebody started such a document.

5. Non-caption use cases

Instead of continuing to look back and improve our support of captions/subtitles in WebVTT, one session at FOMS also went ahead and looked forward to other use cases. The following requirements came out of this:

5.1 Preview Thumbnails

A common use case for timed data is the use of preview thumbnails on the navigation bar of videos. A native implementation of preview thumbnails would allow crawlers and search engines to have a standardised way of extracting timed images for media files, so introduction of a new @kind value “thumbnails” was suggested.

The content of a “thumbnails” cue could be any of:

  • an image URL
  • a sprite URL to a single image
  • a spatial & temporal media fragment URL to a media resource
  • base64 encoded image (data URI)
  • an iframe offset to the media resource

The suggestion is to allow anything that would work in a img @src attribute as value in a cue of @kind=”thumbnails”. Responsive images might also be useful for a track of @kind=”thumbnails”. It may even be possible to define an inband thumbnail track based on the track of @kind=”thumbnails”. Such cues should also work in the JavaScript track API.

5.2 Chapter markers

There is interest to put richer content than just a chapter title into chapter cues. Often, chapters consist of a title, text and and image. The text is not so important, but the image is used almost everywhere that chapters are used. There may be a need to extend chapter cue content with images, similar to what a @kind=”thumbnails” track offers.

The conclusion that we arrived at was that we need to make @kind=”thumbnails” work first and then look at using the learnings from that to extend @kind=”chapters”.

5.3 Inband tracks for live video

A difficult topic was opened with the question of how to transport text tracks in live video. In live captioning, end times are never created for cues, but are implied by the start time of the next cue. This is a use case that hasn’t been addressed in HTML5/WebVTT yet. An old proposal to allow a special end time value of “NEXT” was discussed and recommended for adoption. Also, there was support for the spec change that stops blocking loading VTT until all cues have been loaded.

5.4 Cross-domain VTT loading

A brief discussion centered around the fact that the spec disallows cross-domain loading of WebVTT files, but that no browser implements this. This needs to be discussion at the HTML WG level.

6. Regions in live captioning

The final topic that we discussed was how we could provide support for regions in live captioning.

  • The currently active region definitions will need to be come part of every header of every VTT file segment that HLS uses, so it’s available in case the cues in the segment file reference it.
  • “NEXT” in end time markers would make authoring of live captioned VTT files easier.
  • If the application wants to use 1 word at a time and doesn’t want to delay sending the word until the full cue is authored (e.g. in a Hangout type environment), we will need to introduce the concept of “cue continuation markers”, so we know that a cue could be extended with the next VTT file fragment.

This is an extensive and impressive amount of discussion around WebVTT and a lot of new work to be performed in the future. I’m very grateful for all the people who have contributed to these discussions at FOMS and will hopefully continue to help get the specifications right.

WebVTT as a W3C Recommendation

Three weeks ago I attended TPAC, the annual meeting of W3C Working Groups. One of the meetings was of the Timed Text Working Group (TT-WG), that has been specifying TTML, the Timed Text Markup Language. It is now proposed that WebVTT be also standardised through the same Working Group.

How did that happen, you may ask, in particular since WebVTT and TTML have in the past been portrayed as rival caption formats? How will the WebVTT spec that is currently under development in the Text Track Community Group (TT-CG) move through a Working Group process?

I’ll explain first why there is a need for WebVTT to become a W3C Recommendation, and then how this is proposed to be part of the Timed Text Working Group deliverables, and finally how I can see this working between the TT-CG and the TT-WG.

Advantages of a W3C Recommendation

TTML is a XML-based markup format for captions developed during the time that XML was all the hotness. It has become a W3C standard (a so-called “Recommendation”) despite not having been implemented in any browsers (if you ask me: that’s actually a flaw of the W3C standardisation process: it requires only two interoperable implementations of any kind – and that could be anyone’s JavaScript library or Flash demonstrator – it doesn’t actually require browser implementations. But I digress…). To be fair, a subpart of TTML is by now implemented in Internet Explorer, but all the other major browsers have thus far rejected proposals of implementation.

Because of its Recommendation status, TTML has become the basis for several other caption standards that other SDOs have picked: the SMPTE’s SMPTE-TT format, the EBU’s EBU-TT format, and the DASH Industry Forum’s use of SMPTE-TT. SMPTE-TT has also become the “safe harbour” format for the US legislation on captioning as decided by the FCC. (Note that the FCC requirements for captions on the Web are actually based on a list of features rather than requiring a specific format. But that will be the topic of a different blog post…)

WebVTT is much younger than TTML. TTML was developed as an interchange format among caption authoring systems. WebVTT was built for rendering in Web browsers and with HTML5 in mind. It meets the requirements of the <track> element and supports more than just captions/subtitles. WebVTT is popular with browser developers and has already been implemented in all major browsers (Firefox Nightly is the last to implement it – all others have support already released).

As we can see and as has been proven by the HTML spec and multiple other specs: browsers don’t wait for specifications to have W3C Recommendation status before they implement them. Nor do they really care about the status of a spec – what they care about is whether a spec makes sense for the Web developer and user communities and whether it fits in the Web platform. WebVTT has obviously achieved this status, even with an evolving spec. (Note that the spec tries very hard not to break backwards compatibility, thus all past implementations will at least be compatible with the more basic features of the spec.)

Given that Web browsers don’t need WebVTT to become a W3C standard, why then should we spend effort in moving the spec through the W3C process to become a W3C Recommendation?

The modern Web is now much bigger than just Web browsers. Web specifications are being used in all kinds of devices including TV set-top boxes, phone and tablet apps, and even unexpected devices such as white goods. Videos are increasingly omnipresent thus exposing deaf and hard-of-hearing users to ever-growing challenges in interacting with content on diverse devices. Some of these devices will not use auto-updating software but fixed versions so can’t easily adapt to new features. Thus, caption producers (both commercial and community) need to be able to author captions (and other video accessibility content as defined by the HTML5 element) towards a feature set that is clearly defined to be supported by such non-updating devices.

Understandably, device vendors in this space have a need to build their technology on standardised specifications. SDOs for such device technologies like to reference fixed specifications so the feature set is not continually updating. To reference WebVTT, they could use a snapshot of the specification at any time and reference that, but that’s not how SDOs work. They prefer referencing an officially sanctioned and tested version of a specification – for a W3C specification that means creating a W3C Recommendation of the WebVTT spec.

Taking WebVTT on a W3C recommendation track is actually advantageous for browsers, too, because a test suite will have to be developed that proves that features are implemented in an interoperable manner. In summary, I can see the advantages and personally support the effort to take WebVTT through to a W3C Recommendation.

Choice of Working Group

FAIK this is the first time that a specification developed in a Community Group is being moved into the recommendation track. This is something that has been expected when the W3C created CGs, but not something that has an established process yet.

The first question of course is which WG would take it through to Recommendation? Would we create a new Working Group or find an existing one to move the specification through? Since WGs involve a lot of overhead, the preference was to add WebVTT to the charter of an existing WG. The two obvious candidates were the HTML WG and the TT-WG – the first because it’s where WebVTT originated and the latter because it’s the closest thematically.

Adding a deliverable to a WG is a major undertaking. The TT-WG is currently in the process of re-chartering and thus a suggestion was made to add WebVTT to the milestones of this WG. TBH that was not my first choice. Since I’m already an editor in the HTML WG and WebVTT is very closely related to HTML and can be tested extensively as part of HTML, I preferred the HTML WG. However, adding WebVTT to the TT-WG has some advantages, too.

Since TTML is an exchange format, lots of captions that will be created (at least professionally) will be in TTML and TTML-related formats. It makes sense to create a mapping from TTML to WebVTT for rendering in browsers. The expertise of both, TTML and WebVTT experts is required to develop a good mapping – as has been shown when we developed the mapping from CEA608/708 to WebVTT. Also, captioning experts are already in the TT-WG, so it helps to get a second set of eyes onto WebVTT.

A disadvantage of moving a specification out of a CG into a WG is, however, that you potentially lose a lot of the expertise that is already involved in the development of the spec. People don’t easily re-subscribe to additional mailing lists or want the additional complexity of involving another community (see e.g. this email).

So, a good process needs to be developed to allow everyone to contribute to the spec in the best way possible without requiring duplicate work. How can we do that?

The forthcoming process

At TPAC the TT-WG discussed for several hours what the next steps are in taking WebVTT through the TT-WG to recommendation status (agenda with slides). I won’t bore you with the different views – if you are keen, you can read the minutes.

What I came away with is the following process:

  1. Fix a few more bugs in the CG until we’re happy with the feature set in the CG. This should match the feature set that we realistically expect devices to implement for a first version of the WebVTT spec.
  2. Make a FSA (Final Specification Agreement) in the CG to create a stable reference and a clean IPR position.
  3. Assuming that the TT-WG’s charter has been approved with WebVTT as a milestone, we would next bring the FSA specification into the TT-WG as FPWD (First Public Working Draft) and immediately do a Last Call which effectively freezes the feature set (this is possible because there has already been wide community review of the WebVTT spec); in parallel, the CG can continue to develop the next version of the WebVTT spec with new features (just like it is happening with the HTML5 and HTML5.1 specifications).
  4. Develop a test suite and address any issues in the Last Call document (of course, also fix these issues in the CG version of the spec).
  5. As per W3C process, substantive and minor changes to Last Call documents have to be reported and raised issues addressed before the spec can progress to the next level: Candidate Recommendation status.
  6. For the next step – Proposed Recommendation status – an implementation report is necessary, and thus the test suite needs to be finalized for the given feature set. The feature set may also be reduced at this stage to just the ones implemented interoperably, leaving any other features for the next version of the spec.
  7. The final step is Recommendation status, which simply requires sufficient support and endorsement by W3C members.

The first version of the WebVTT spec naturally has a focus on captioning (and subtitling), since this has been the dominant use case that we have focused on this far and it’s the part that is the most compatibly implemented feature set of WebVTT in browsers. It’s my expectation that the next version of WebVTT will have a lot more features related to audio descriptions, chapters and metadata. Thus, this seems a good time for a first version feature freeze.

There are still several obstacles towards progressing WebVTT as a milestone of the TT-WG. Apart from the need to get buy-in from the TT-WG, the TT-CG, and the AC (Adivisory Committee who have to approve the new charter), we’re also looking at the license of the specification document.

The CG specification has an open license that allows creating derivative work as long as there is attribution, while the W3C document license for documents on the recommendation track does not allow the creation of derivative work unless given explicit exceptions. This is an issue that is currently being discussed in the W3C with a proposal for a CC-BY license on the Recommendation track. However, my view is that it’s probably ok to use the different document licenses: the TT-WG will work on WebVTT 1.0 and give it a W3C document license, while the CG starts working on the next WebVTT version under the open CG license. It probably actually makes sense to have a less open license on a frozen spec.

Making the best of a complicated world

WebVTT is now proposed as part of the recharter of the TT-WG. I have no idea how complicated the process will become to achieve a W3C WebVTT 1.0 Recommendation, but I am hoping that what is outlined above will be workable in such a way that all of us get to focus on progressing the technology.

At TPAC I got the impression that the TT-WG is committed to progressing WebVTT to Recommendation status. I know that the TT-CG is committed to continue developing WebVTT to its full potential for all kinds of media-time aligned content with new kinds already discussed at FOMS. Let’s enable both groups to achieve their goals. As a consequence, we will allow the two formats to excel where they do: TTML as an interchange format and WebVTT as a browser rendering format.

Summary Video Accessibility Talk

I’ve just got off a call to the UK Digital TV Group, for which I gave a talk on HTML5 video accessibility (slides best viewed in Google Chrome).

The slide provide a high-level summary of the accessibility features that we’ve developed in the W3C for HTML5, including:

  • Subtitles & Captions with WebVTT and the track element
  • Video Descriptions with WebVTT, the track element and speech synthesis
  • Chapters with WebVTT for semantic navigation
  • Audio Descriptions through synchronising an audio track with a video
  • Sign Language video synchronized with a main video

I received some excellent questions.

The obvious one was about why WebVTT and not TTML. While for anyone who has tried to implement TTML support, the advantages of WebVTT should be clear, for some the decision of the browsers to go with WebVTT still seems to be bothersome. The advantages of CSS over XSL-FO in a browser-context are obvious, but not as much outside browsers. So, the simplicity of WebVTT and the clear integration with HTML have to speak for themselves. Conversion between TTML and WebVTT was a feature that was being asked for.

I received a question about how to support ducking (reduce the volume of the main audio track) when using video descriptions. My reply was to either use video descriptions with WebVTT and do ducking during the times that a cue is active, or when using audio descriptions (i.e. actual audio tracks) to add an additional WebVTT file of kind=metadata to mark the intervals in which to do ducking. In both cases some JavaScript will be necessary.

I received another question about how to do clean audio, which I had almost forgotten was a requirement from our earlier media accessibility document. “Clean audio” consists of isolating the audio channel containing the spoken dialog and important non-speech information that can then be amplified or otherwise modified, while other channels containing music or ambient sounds are attenuated. I suggested using the mediagroup attribute to provide a main video element (without an audio track) and then the other channels as parallel audio tracks that can be turned on and off and attenuated individually. There is some JavaScript coding involved on top of the APIs that we have defined in HTML, but it can be implemented in browsers that support the mediagroup attribute.

Another question was about the possibilities to extend the list of @kind attribute values. I explained that right now we have a proposal for a new text track kind=”forced” so as to provide forced subtitles for sections of video with foreign language. These would be on when no other subtitle or caption tracks are activated. I also explained that if there is a need for application-specific text tracks, the kind=”metadata” would be the correct choice.

I received some further questions, in particular about how to apply styling to captions (e.g. color changes to text) and about how closely the browser are able to keep synchronization across multiple media elements. The earlier was easily answered with the ::cue pseudo-element, but the latter is a quality of implementation feature, so I had to defer to individual browsers.

Overall it was a good exercise to summarize the current state of HTML5 video accessibility and I was excited to show off support in Chrome for all the features that we designed into the standard.

New Challenges

I finished up at Google last week and am now working at NICTA, an Australian ICT research institute.

My work with Google was exciting and I learned a lot. I like to think that Google also got a lot out of me – I coded and contributed to some YouTube caption features, I worked on Chrome captions and video controls, and above all I worked on video accessibility for HTML at the W3C.

I was one of the key authors of the W3C Media Accessibility Requirements document that we created in the Media Accessibility Task Force of the W3C HTML WG. I then went on to help make video accessibility a reality. We created WebVTT and the <track> element and applied it to captions, subtitles, chapters (navigation), video descriptions, and metadata. To satisfy the need for synchronisation of video with other media resources such as sign language video or audio descriptions, we got the MediaController object and the @mediagroup attribute.

I must say it was a most rewarding time. I learned a lot about being productive at Google, collaborate successfully over the distance, about how the WebKit community works, and about the new way of writing W3C standard (which is more like pseudo-code). As one consequence, I am now a co-editor of the W3C HTML spec and it seems I am also about to become the editor of the WebVTT spec.

At NICTA my new focus of work is WebRTC. There is both a bit of research and a whole bunch of application development involved. I may even get to do some WebKit development, if we identify any issues with the current implementation. I started a week ago and am already amazed by the amount of work going on in the WebRTC space and the amazing number of open source projects playing around with it. Video conferencing is a new challenge and I look forward to it.

Video Conferencing in HTML5: WebRTC via Socket.io

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.

The use cases for a element in HTML

The W3C HTML WG and the WHATWG are currently discussing the introduction of a <main> element into HTML.

The <main> element has been proposed by Steve Faulkner and is specified in a draft extension spec which is about to be accepted as a FPWD (first public working draft) by the W3C HTML WG. This implies that the W3C HTML WG will be looking for implementations and for feedback by implementers on this spec.

I am supportive of the introduction of a <main> element into HTML. However, I believe that the current spec and use case list don’t make a good enough case for its introduction. Here are my thoughts.

Main use case: accessibility

In my opinion, the main use case for the introduction of <main> is accessibility.

Like any other users, when blind users want to perceive a Web page/application, they need to have a quick means of grasping the content of a page. Since they cannot visually scan the layout and thus determine where the main content is, they use accessibility technology (AT) to find what is known as “landmarks”.

“Landmarks” tell the user what semantic content is on a page: a header (such as a banner), a search box, a navigation menu, some asides (also called complementary content), a footer, …. and the most important part: the main content of the page. It is this main content that a blind user most often wants to skip to directly.

In the days of HTML4, a hidden “skip to content” link at the beginning of the Web page was used as a means to help blind users access the main content.

In the days of ARIA, the aria @role=main enables authors to avoid a hidden link and instead mark the element where the main content begins to allow direct access to the main content. This attribute is supported by AT – in particular screen readers – by making it part of the landmarks that AT can directly skip to.

Both the hidden link and the ARIA @role=main approaches are, however, band aids: they are being used by those of us that make “finished” Web pages accessible by adding specific extra markup.

A world where ARIA is not necessary and where accessibility developers would be out of a job because the normal markup that everyone writes already creates accessible Web sites/applications would be much preferable over the current world of band-aids.

Therefore, to me, the primary use case for a <main> element is to achieve exactly this better world and not require specialized markup to tell a user (or a tool) where the main content on a page starts.

An immediate effect would be that pages that have a <main> element will expose a “main” landmark to blind and vision-impaired users that will enable them to directly access that main content on the page without having to wade through other text on the page. Without a <main> element, this functionality can currently only be provided using heuristics to skip other semantic and structural elements and is for this reason not typically implemented in AT.

Other use cases

The <main> element is a semantic element not unlike other new semantic elements such as <header>, <footer>, <aside>, <article>, <nav>, or <section>. Thus, it can also serve other uses where the main content on a Web page/Web application needs to be identified.

Data mining

For data mining of Web content, the identification of the main content is one of the key challenges. Many scholarly articles have been published on this topic. This stackoverflow article references and suggests a multitude of approaches, but the accepted answer says “there’s no way to do this that’s guaranteed to work”. This is because Web pages are inherently complex and many <div>, <p>, <iframe> and other elements are used to provide markup for styling, notifications, ads, analytics and other use cases that are necessary to make a Web page complete, but don’t contribute to what a user consumes as semantically rich content. A <main> element will allow authors to pro-actively direct data mining tools to the main content.

Search engines

One particularly important “data mining” tool are search engines. They, too, have a hard time to identify which sections of a Web page are more important than others and employ many heuristics to do so, see e.g. this ACM article. Yet, they still disappoint with poor results pointing to findings of keywords in little relevant sections of a page rather than ranking Web pages higher where the keywords turn up in the main content area. A <main> element would be able to help search engines give text in main content areas a higher weight and prefer them over other areas of the Web page. It would be able to rank different Web pages depending on where on the page the search words are found. The <main> element will be an additional hint that search engines will digest.

Visual focus

On small devices, the display of Web pages designed for Desktop often causes confusion as to where the main content can be found and read, in particular when the text ends up being too small to be readable. It would be nice if browsers on small devices had a functionality (maybe a default setting) where Web pages would start being displayed as zoomed in on the main content. This could alleviate some of the headaches of responsive Web design, where the recommendation is to show high priority content as the first content. Right now this problem is addressed through stylesheets that re-layout the page differently depending on device, but again this is a band-aid solution. Explicit semantic markup of the main content can solve this problem more elegantly.

Styling

Finally, naturally, <main> would also be used to style the main content differently from others. You can e.g. replace a semantically meaningless <div id=”main”> with a semantically meaningful <main> where their position is identical. My analysis below shows, that this is not always the case, since oftentimes <div id=”main”> is used to group everything together that is not the header – in particular where there are multiple columns. Thus, the ease of styling a <main> element is only a positive side effect and not actually a real use case. It does make it easier, however, to adapt the style of the main content e.g. with media queries.

Proposed alternative solutions

It has been proposed that existing markup serves to satisfy the use cases that <main> has been proposed for. Let’s analyse these on some of the most popular Web sites. First let’s list the propsed algorithms.

Proposed solution No 1: Scooby-Doo

On Sat, Nov 17, 2012 at 11:01 AM, Ian Hickson <ian@hixie.ch> wrote:
| The main content is whatever content isn't
| marked up as not being main content (anything not marked up with <header>,
| <aside>, <nav>, etc).

This implies that the first element that is not a <header>, <aside>, <nav>, or <footer> will be the element that we want to give to a blind user as the location where they should start reading. The algorithm is implemented in https://gist.github.com/4032962.

Proposed solution No 2: First article element

On Sat, Nov 17, 2012 at 8:01 AM, Ian Hickson  wrote:
| On Thu, 15 Nov 2012, Ian Yang wrote:
| >
| > That's a good idea. We really need an element to wrap all the <p>s,
| > <ul>s, <ol>s, <figure>s, <table>s ... etc of a blog post.
|
| That's called <article>.

This approach identifies the first <article> element on the page as containing the main content. Here’s the algorithm for this approach.

Proposed solution No 3: An example heuristic approach

The readability plugin has been developed to make Web pages readable by essentially removing all the non-main content from a page. An early source of readability is available. This demonstrates what a heuristic approach can perform.

Analysing alternative solutions

Comparison

I’ve picked 4 typical Websites (top on Alexa) to analyse how these three different approaches fare. Ideally, I’d like to simply apply the above three scripts and compare pictures. However, since the semantic HTML5 elements <header>, <aside>, <nav>, and <footer> are not actually used by any of these Web sites, I don’t actually have this choice.

So, instead, I decided to make some assumptions of where these semantic elements would be used and what the outcome of applying the first two algorithms would be. I can then compare it to the third, which is a product so we can take screenshots.

Google.com

http://google.com – search for “Scooby Doo”.

The search results page would likely be built with:

  • a <nav> menu for the Google bar
  • a <header> for the search bar
  • another <header> for the login section
  • another <nav> menu for the search types
  • a <div> to contain the rest of the page
  • a <div> for the app bar with the search number
  • a few <aside>s for the left and right column
  • a set of <article>s for the search results
“Scooby Doo” would find the first element after the headers as the “main content”. This is the element before the app bar in this case. Interestingly, there is a <div @id=main> already in the current Google results page, which “Scooby Doo” would likely also pick. However, there are a nav bar and two asides in this div, which clearly should not be part of the “main content”. Google actually placed a @role=main on a different element, namely the one that encapsulates all the search results.

“First Article” would find the first search result as the “main content”. While not quite the same as what Google intended – namely all search results – it is close enough to be useful.

The “readability” result is interesting, since it is not able to identify the main text on the page. It is actually aware of this problem and brings a warning before displaying this page:

Readability of google.com

Facebook.com

https://facebook.com

A user page would likely be built with:

  • a <header> bar for the search and login bar
  • a <div> to contain the rest of the page
  • an <aside> for the left column
  • a <div> to contain the center and right column
  • an <aside> for the right column
  • a <header> to contain the center column “megaphone”
  • a <div> for the status posting
  • a set of <article>s for the home stream
“Scooby Doo” would find the first element after the headers as the “main content”. This is the element that contains all three columns. It’s actually a <div @id=content> already in the current Facebook user page, which “Scooby Doo” would likely also pick. However, Facebook selected a different element to place the @role=main : the center column.

“First Article” would find the first news item in the home stream. This is clearly not what Facebook intended, since they placed the @role=main on the center column, above the first blog post’s title. “First Article” would miss that title and the status posting.

The “readability” result again disappoints but warns that it failed:

YouTube.com

http://youtube.com

A video page would likely be built with:

  • a <header> bar for the search and login bar
  • a <nav> for the menu
  • a <div> to contain the rest of the page
  • a <header> for the video title and channel links
  • a <div> to contain the video with controls
  • a <div> to contain the center and right column
  • an <aside> for the right column with an <article> per related video
  • an <aside> for the information below the video
  • a <article> per comment below the video
“Scooby Doo” would find the first element after the headers as the “main content”. This is the element that contains the rest of the page. It’s actually a <div @id=content> already in the current YouTube video page, which “Scooby Doo” would likely also pick. However, YouTube’s related videos and comments are unlikely to be what the user would regard as “main content” – it’s the video they are after, which generously has a <div id=watch-player>.

“First Article” would find the first related video or comment in the home stream. This is clearly not what YouTube intends.

The “readability” result is not quite as unusable, but still very bare:

Wikipedia.com

http://wikipedia.com (“Overscan” page)

A Wikipedia page would likely be built with:

  • a <header> bar for the search, login and menu items
  • a <div> to contain the rest of the page
  • an &ls; article> with title and lots of text
  • <article> an <aside> with the table of contents
  • several <aside>s for the left column
Good news: “Scooby Doo” would find the first element after the headers as the “main content”. This is the element that contains the rest of the page. It’s actually a <div id=”content” role=”main”> element on Wikipedia, which “Scooby Doo” would likely also pick.

“First Article” would find the title and text of the main element on the page, but it would also include an <aside>.

The “readability” result is also in agreement.

Results

In the following table we have summarised the results for the experiments:

Site Scooby-Doo First article Readability
Google.com FAIL SUCCESS FAIL
Facebook.com FAIL FAIL FAIL
YouTube.com FAIL FAIL FAIL
Wikipedia.com SUCCESS SUCCESS SUCCESS

Clearly, Wikipedia is the prime example of a site where even the simple approaches find it easy to determine the main content on the page. WordPress blogs are similarly successful. Almost any other site, including news sites, social networks and search engine sites are petty hopeless with the proposed approaches, because there are too many elements that are used for layout or other purposes (notifications, hidden areas) such that the pre-determined list of semantic elements that are available simply don’t suffice to mark up a Web page/application completely.

Conclusion

It seems that in general it is impossible to determine which element(s) on a Web page should be the “main” piece of content that accessibility tools jump to when requested, that a search engine should put their focus on, or that should be highlighted to a general user to read. It would be very useful if the author of the Web page would provide a hint through a <main> element where that main content is to be found.

I think that the <main> element becomes particularly useful when combined with a default keyboard shortcut in browsers as proposed by Steve: we may actually find that non-accessibility users will also start making use of this shortcut, e.g. to get to videos on YouTube pages directly without having to tab over search boxes and other interactive elements, etc. Worthwhile markup indeed.

What is “interoperable TTML”?

I’ve just tried to come to terms with the latest state of TTML, the Timed Text Markup Language.

TTML has been specified by the W3C Timed Text Working Group and released as a RECommendation v1.0 in November 2010. Since then, several organisations have tried to adopt it as their caption file format. This includes the SMPTE, the EBU (European Broadcasting Union), and Microsoft.

Both, Microsoft and the EBU actually looked at TTML in detail and decided that in order to make it usable for their use cases, a restriction of its functionalities is needed.

EBU-TT

The EBU released EBU-TT, which restricts the set of valid attributes and feature. “The EBU-TT format is intended to constrain the features provided by TTML, especially to make EBU-TT more suitable for the use with broadcast video and web video applications.” (see EBU-TT).

In addition, EBU-specific namespaces were introduce to extend TTML with EBU-specific data types, e.g. ebuttdt:frameRateMultiplierType or ebuttdt:smpteTimingType. Similarly, a bunch of metadata elements were introduced, e.g. ebuttm:documentMetadata, ebuttm:documentEbuttVersion, or ebuttm:documentIdentifier.

The use of namespaces as an extensibility mechanism will ascertain that EBU-TT files continue to be valid TTML files. However, any vanilla TTML parser will not know what to do with these custom extensions and will drop them on the floor.

Simple Delivery Profile

With the intention to make TTML ready for “internet delivery of Captions originated in the United States”, Microsoft proposed a “Simple Delivery Profile for Closed Captions (US)” (see Simple Profile). The Simple Profile is also a restriction of TTML.

Unfortunately, the Microsoft profile is not the same as the EBU-TT profile: for example, it contains the “set” element, which is not conformant in EBU-TT. Similarly, the supported style features are different, e.g. Simple Profile supports “display-region”, while EBU-TT does not. On the other hand, EBU-TT supports monospace, sans-serif and serif fonts, while the Simple profile does not.

Thus files created for the Simple Delivery Profile will not work on players that expect EBU-TT and the reverse.

Fortunately, the Simple Delivery Profile does not introduce any new namespaces and new features, so at least it is an explicit subpart of TTML and not both a restriction and extension like EBU-TT.

SMPTE-TT

SMPTE also created a version of the TTML standard called SMPTE-TT. SMPTE did not decide on a subset of TTML for their purposes – it was simply adopted as a complete set. “This Standard provides a framework for timed text to be supported for content delivered via broadband means,…” (see SMPTE-TT).

However, SMPTE extended TTML in SMPTE-TT with an ability to store a binary blob with captions in another format. This allows using SMPTE-TT as a transport format for any caption format and is deemed to help with “backwards compatibility”.

Now, instead of specifying a profile, SMPTE decided to define how to convert CEA-608 captions to SMPTE-TT. Even if it’s not called a “profile”, that’s actually what it is. It even has its own namespace: “m608:”.

Conclusion

With all these different versions of TTML, I ask myself what a video player that claims support for TTML will do to get something working. The only chance it has is to implement all the extensions defined in all the different profiles. I pity the player that has to deal with a SMPTE-TT file that has a binary blob in it and is expected to be able to decode this.

Now, what is a caption author supposed to do when creating TTML? They obviously cannot expect all players to be able to play back all TTML versions. Should they create different files depending on what platform they are targeting, i.e. a EBU-TT version, a SMPTE-TT version, a vanilla TTML version, and a Simple Delivery Profile version? Should they by throwing all the features of all the versions into one TTML file and hope that the players will pick out the right things that they require and drop the rest on the floor?

Maybe the best way to progress would be to make a list of the “safe” features: those features that every TTML profile supports. That may be the best way to get an “interoperable TTML” file. Here’s me hoping that this minimal set of features doesn’t just end up being the usual (starttime, endtime, text) triple.

UPDATE:

I just found out that UltraViolet have their own profile of SMPTE-TT called CFF-TT (see UltraViolet FAQ and spec). They are making some SMPTE-TT fields optional, but introduce a new @forcedDisplayMode attribute under their own namespace “cff:”.