Category Archives: FOMS

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 ‎ and ‏ 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.

Open Media Developers Track at OVC 2011

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

The new FOMS: Open Media Developers at OVC

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!

adaptive HTTP streaming for open codecs

At this week’s FOMS in New York we had one over-arching topic that seemed to be of interest to every single participant: how to do adaptive bitrate streaming over HTTP for open codecs. On the first day, there was a general discussion about the advantages and disadvantages of adaptive HTTP streaming, while on the second day, we moved towards designing a solution for Ogg and WebM. While I didn’t attend all the discussions, I want to summarize the insights that I took out of the days in this blog post and the alternative implementation strategies that were came up with.

Use Cases for Adaptive HTTP Streaming

Streaming using RTP/RTSP has in the past been the main protocol to provide live video streams, either for broadcast or for real-time communication. It has been purpose-built for chunked video delivery and has features that many customers want, such as the ability to encrypt the stream, to tell players not to store the data, and to monitor the performance of the stream such that its bandwidth can be adapted. It has, however, also many disadvantages, not least that it goes over ports that normal firewalls block and thus is rather difficult to deploy, but also that it requires special server software, a client that speaks the protocol, and has a signalling overhead on the transport layer for adapting the stream.

RTP/RTSP has been invented to allow for high quality of service video consumption. In the last 10 years, however, it has become the norm to consume “canned” video (i.e. non-live video) over HTTP, making use of the byte-range request functionality of HTTP for seeking. While methods have been created to estimate the size of a pre-buffer before starting to play back in order to achieve continuous playback based on the bandwidth of your pipe at the beginning of downloading, not much can be done when one runs out of pre-buffer in the middle of playback or when the CPU on the machine doesn’t manage to catch up with decoding of the sheer amount of video data: your playback stops to go into re-buffering in the first case and starts to become choppy in the latter case.

An obvious approach to improving this situation is the scale the bandwidth of the video stream down, potentially even switch to a lower resolution video, right in the middle of playback. Apple’s HTTP live streaming, Microsoft’s Smooth Streaming, and Adobe’s Dynamic Streaming are all solutions in this space. Also, ISO/MPEG is working on DASH (Dynamic Adaptive Streaming over HTTP) is an effort to standardize the approach for MPEG media. No solution yets exist for the open formats within Ogg or WebM containers.

Some features of HTTP adaptive streaming are:

  • Enables adaptation of downloading to avoid continuing buffering when network or machine cannot cope.
  • Gapless switching between streams of different bitrate.
  • No special server software is required – any existing Web Server can be used to provide the streams.
  • The adaptation comes from the media player that actually knows what quality the user experiences rather than the network layer that knows nothing about the performance of the computer, and can only tell about the performance of the network.
  • Adaptation means that several versions of different bandwidth are made available on the server and the client switches between them based on knowledge it has about the video quality that the user experiences.
  • Bandwidth is not wasted by downloading video data that is not being consumed by the user, but rather content is pulled moments just before it is required, which works both for the live and canned content case and is particularly useful for long-form content.


In discussions at FOMS it was determined that mid-stream switching between different bitrate encoded audio files is possible. Just looking at the PCM domain, it requires stitching the waveform together at the switch-over point, but that is not a complex function. To be able to do that stitching with Vorbis-encoded files, there is no need for a overlap of data, because the encoded samples of the previous window in a different bitrate page can be used as input into the decoding of the current bitrate page, as long as the resulting PCM samples are stitched.

For video, mid-stream switching to a different bitrate encoded stream is also acceptable, as long as the switch-over point adheres to a keyframe, which can be independently decoded.

Thus, the preparation of the alternative bitstream videos requires temporal synchronisation of keyframes on video – the audio can deal with the switch-over at any point. A bit of intelligent encoding is thus necessary – requiring the encoding pipeline to provide regular keyframes at a certain rate would be sufficient. Then, the switch-over points are the keyframes.

Technical Realisation

With the solutions from Adobe, Microsoft and Apple, the technology has been created such there are special tools on the server that prepare the content for adaptive HTTP streaming and provide a manifest of the prepared content. Typically, the content is encoded in versions of different bitrates and the bandwidth versions are broken into chunks that can be decoded independently. These chunks are synchronised between the different bitrate versions such that there are defined switch-over points. The switch-over points as well as the file names of the different chunks are documented inside a manifest file. It is this manifest file that the player downloads instead of the resource at the beginning of streaming. This manifest file informs the player of the available resources and enables it to orchestrate the correct URL requests to the server as it progresses through the resource.

At FOMS, we took a step back from this approach and analysed what the general possibilities are for solving adaptive HTTP streaming. For example, it would be possible to not chunk the original media data, but instead perform range requests on the different bitrate versions of the resource. The following options were identified.


With Chunking, the original bitrate versions are chunked into smaller full resources with defined switch-over points. This implies creation of a header on each one of the chunks and thus introduces overhead. Assuming we use 10sec chunks and 6kBytes per chunk, that results in 5kBit/sec extra overhead. After chunking the files this way, we provide a manifest file (similar to Apple’s m3u8 file, or the SMIL-based manifest file of Microsoft, or Adobe’s Flash Media Manifest file). The manifest file informs the client about the chunks and the switch-over points and the client requests those different resources at the switch-over points.


  • Header overhead on the pipe.
  • Switch-over delay for decoding the header.
  • Possible problem with TCP slowstart on new files.
  • A piece of software is necessary on server to prepare the chunked files.
  • A large amount of files to manage on the server.
  • The client has to hide the switching between full resources.


  • Works for live streams, where increasing amounts of chunks are written.
  • Works well with CDNs, because mid-stream switching to another server is easy.
  • Chunks can be encoded such that there is no overlap in the data necessary on switch-over.
  • May work well with Web sockets.
  • Follows the way in which proprietary solutions are doing it, so may be easy to adopt.
  • If the chunks are concatenated on the client, you get chained Ogg files (similar concept in WebM?), which are planned to be supported by Web browsers and are thus legal files.

Chained Chunks

Alternatively to creating the large number of files, one could also just create the chained files. Then, the switch-over is not between different files, but between different byte ranges. The headers still have to be read and parsed. And a manifest file still has to exist, but it now points to byte ranges rather than different resources.

Advantages over Chunking:

  • No TCP-slowstart problem.
  • No large number of files on the server.

Disadvantages over Chunking:

  • Mid-stream switching to other servers is not easily possible – CDNs won’t like it.
  • Doesn’t work with Web sockets as easily.
  • New approach that vendors will have to grapple with.

Virtual Chunks

Since in Chained Chunks we are already doing byte-range requests, it is a short step towards simply dropping the repeating headers and just downloading them once at the beginning for all possible bitrate files. Then, as we seek to different positions in “the” file, the byte range of the bitrate version that makes sense to retrieve at that stage would be requested. This could even be done with media fragment URIs, through addressing with time ranges is less accurate than explicit byte ranges.

In contrast to the previous two options, this basically requires keeping n different encoding pipelines alive – one for every bitrate version. Then, the byte ranges of the chunks will be interpreted by the appropriate pipeline. The manifest now points to keyframes as switch-over points.

Advantage over Chained Chunking:

  • No header overhead.
  • No continuous re-initialisation of decoding pipelines.

Disadvantages over Chained Chunking:

  • Multiple decoding pipelines need to be maintained and byte ranges managed for each.

Unchunked Byte Ranges

We can even consider going all the way and not preparing the alternative bitrate resources for switching, i.e. not making sure that the keyframes align. This will then require the player to do the switching itself, determine when the next keyframe comes up in its current stream then seek to that position in the next stream, always making sure to go back to the last keyframe before that position and discard all data until it arrives at the same offset.


  • There will be an overlap in the timeline for download, which has to be managed from the buffering and alignment POV.
  • Overlap poses a challenge of downloading more data than necessary at exactly the time where one doesn’t have bandwidth to spare.
  • Requires seeking.
  • Messy.


  • No special authoring of resources on the server is needed.
  • Requires a very simple manifest file only with a list of alternative bitrate files.

Final concerns

At FOMS we weren’t able to make a final decision on how to achieve adaptive HTTP streaming for open codecs. Most agreed that moving forward with the first case would be the right thing to do, but the sheer number of files that can create is daunting and it would be nice to avoid that for users.

Other goals are to make it work in stand-alone players, which means they will need to support loading the manifest file. And finally we want to enable experimentation in the browser through JavaScript implementation, which means there needs to be an interface to provide the quality of decoding to JavaScript. Fortunately, a proposal for such a statistics API already exists. The number of received frames, the number of dropped frames, and the size of the video are the most important statistics required.

Upcoming conferences / workshops

Lots is happening in open source multimedia land in the next few months.

Check out these cool upcoming conferences / workshops / miniconfs…

September 29th and 30th, New York
Open Subtitles Design Summit

October 1st and 2nd, New York
Open Video Conference

October 3rd and 4th, New York
Foundations of Open Media Software Developer Workshop

January 24/25th, Brisbane, Australia
LCA Multimedia Miniconf

Accessibility support in Ogg and liboggplay

At the recent FOMS/LCA in Wellington, New Zealand, we talked a lot about how Ogg could support accessibility. Technically, this means support for multiple text tracks (subtitles/captions), multiple audio tracks (audio descriptions parallel to main audio track), and multiple video tracks (sign language video parallel to main video track).

Creating multitrack Ogg files
The creation of multitrack Ogg files is already possible using one of the muxing applications, e.g. oggz-merge. For example, I have my own little collection of multitrack Ogg files at But then you are stranded with files that no player will play back.

Multitrack Ogg in Players
As Ogg is now being used in multiple Web browsers in the new HTML5 media formats, there are in particular requirements for accessibility support for the hard-of-hearing and vision-impaired. Either multitrack Ogg needs to become more of a common case, or the association of external media files that provide synchronised accessibility data (captions, audio descriptions, sign language) to the main media file needs to become a standard in HTML5.

As it turn out, both these approaches are being considered and worked on in the W3C. Accessibility data that are audio or video tracks will in the near future have to come out of the media resource itself, but captions and other text tracks will also be available from external associated elements.

The availability of internal accessibility tracks in Ogg is a new use case – something Ogg has been ready to do, but has not gone into common usage. MPEG files on the other hand have for a long time been used with internal accessibility tracks and thus frameworks and players are in place to decode such tracks and do something sensible with them. This is not so much the case for Ogg.

For example, a current VLC build installed on Windows will display captions, because Ogg Kate support is activated. A current VLC build on any other platform, however, has Ogg Kate support deactivated in the build, so captions won’t display. This will hopefully change soon, but we have to look also beyond players and into media frameworks – in particular those that are being used by the browser vendors to provide Ogg support.

Multitrack Ogg in Browsers
Hopefully gstreamer (which is what Opera uses for Ogg support) and ffmpeg (which is what Chrome uses for Ogg support) will expose all available tracks to the browser so they can expose them to the user for turning on and off. Incidentally, a multitrack media JavaScript API is in development in the W3C HTML5 Accessibility Task Force for allowing such control.

The current version of Firefox uses liboggplay for Ogg support, but liboggplay’s multitrack support has been sketchy this far. So, Viktor Gal – the liboggplay maintainer – and I sat down at FOMS/LCA to discuss this and Viktor developed some patches to make the demo player in the liboggplay package, the glut-player, support the accessibility use cases.

I applied Viktor’s patch to my local copy of liboggplay and I am very excited to show you the screencast of glut-player playing back a video file with an audio description track and an English caption track all in sync:


Further developments
There are still important questions open: for example, how will a player know that an audio description track is to be played together with the main audio track, but a dub track (e.g. a German dub for an English video) is to be played as an alternative. Such metadata for the tracks is something that Ogg is still missing, but that Ogg can be extended with fairly easily through the use of the Skeleton track. It is something the Xiph community is now working on.

This is great progress towards accessibility support in Ogg and therefore in Web browsers. And there is more to come soon.

Audio Track Accessibility for HTML5

I have talked a lot about synchronising multiple tracks of audio and video content recently. The reason was mainly that I foresee a need for more than two parallel audio and video tracks, such as audio descriptions for the vision-impaired or dub tracks for internationalisation, as well as sign language tracks for the hard-of-hearing.

It is almost impossible to introduce a good scheme to deliver the right video composition to a target audience. Common people will prefer bare a/v, vision-impaired would probably prefer only audio plus audio descriptions (but will probably take the video), and the hard-of-hearing will prefer video plus captions and possibly a sign language track . While it is possible to dynamically create files that contain such tracks on a server and then deliver the right composition, implementation of such a server method has not been very successful in the last years and it would likely take many years to roll out such new infrastructure.

So, the only other option we have is to synchronise completely separate media resource together as they are selected by the audience.

It is this need that this HTML5 accessibility demo is about: Check out the demo of multiple media resource synchronisation.

I created a Ogg video with only a video track (10m53s750). Then I created an audio track that is the original English audio track (10m53s696). Then I used a Spanish dub track that I found through BlenderNation as an alternative audio track (10m58s337). Lastly, I created an audio description track in the original language (10m53s706). This creates a video track with three optional audio tracks.

I took away all native controls from these elements when using the HTML5 audio and video tag and ran my own stop/play and seeking approaches, which handled all media elements in one go.

I was mostly interested in the quality of this experience. Would the different media files stay mostly in sync? They are normally decoded in different threads, so how big would the drift be?

The resulting page is the basis for such experiments with synchronisation.

The page prints the current playback position in all of the media files at a constant interval of 500ms. Note that when you pause and then play again, I am re-synching the audio tracks with the video track, but not when you just let the files play through.

I have let the files play through on my rather busy Macbook and have achieved the following interesting drift over the course of about 9 minutes:

Drift between multiple parallel played media elements

You will see that the video was the slowest, only doing roughly 540s, while the Spanish dub did 560s in the same time.

To fix such drifts, you can always include regular re-synchronisation points into the video playback. For example, you could set a timeout on the playback to re-sync every 500ms. Within such a short time, it is almost impossible to notice a drift. Don’t re-load the video, because it will lead to visual artifacts. But do use the video’s currentTime to re-set the others. (UPDATE: Actually, it depends on your situation, which track is the best choice as the main timeline. See also comments below.)

It is a workable way of associating random numbers of media tracks with videos, in particular in situations where the creation of merged files cannot easily be included in a workflow.

Video Streaming from

You probably heard it already: is live streaming its video in a Microsoft proprietary format.

Fortunately, there is now a re-broadcast that you can get in an open format from . It comes from a server in Europe, but relies on transcoding here in New Zealand, so it may not be completely reliable.

UPDATE: A second server is now also available from the US at

Today, the down under open source / Linux conference in Wellington started with the announcement that every talk and mini-conf will be live streamed to the Internet and later published online. That’s an awesome achievement!

However, minutes after the announcement, I was very disappointed to find out that the streams are actually provided in a proprietary format and through a proprietary streaming protocol: a Microsoft streaming service that provides Windows media streams.

Why stream an open source conference in a proprietary format with proprietary software? If we cannot use our own technologies for our own conferences, how will we get the rest of the world to use them?

I must say, I am personally embarrassed, because I was part of several audio/video teams of previous LCAs that have managed to record and stream content in open formats and with open media software. I would have helped get this going, but wasn’t aware of the situation.

I am also the main organiser of the FOMS Workshop (Foundations of Open Media Software) that ran the week before LCA and brought some of the core programmers in open media software into Wellington, most of which are also attending LCA. We have the brains here and should be able to get this going.

Fortunately, the published content will be made available in Ogg Theora/Vorbis. So, it’s only the publicly available stream that I am concerned about.

Speaking with the organisers, I can somewhat understand how this came to be. They took the “easy” way of delegating the video work to an external company. Even though this company is an expert in open source and networking, their media streaming customers are all using Flash or Windows media software, which are current de-facto standards and provide extra features such as DRM. It seems apart from there were no requests on them for streaming Ogg Theora/Vorbis yet. Their existing infrastructure includes CDN distribution and CDN providers certainly typically don’t provide Ogg Theora/Vorbis support or Icecast streaming.

So, this is actually a problem founded in setting up streaming through a professional service rather than through the community. The way in which this was set up at other events was to get together a group of volunteers that provided streaming reflectors for free. In this way, a community-created CDN is built that can deal with the streams. That there are no professional CDN providers available yet that provide Icecast support is a sign that there is a gap in the market.

But phear not – a few of the FOMS folk got together to fix the situation.

It involved setting up Icecast streams for each room’s video stream. Since there is no access to the raw video stream, there is a need to transcode the video from proprietary codecs to the open Ogg Theora/Vorbis format.

To do this legally, a purchase of the codec libraries from Fluendo was necessary, which cost a whopping EURO 28 and covers all the necessary patent licenses. The glue to get the videos from mms to icecast streams is a GStreamer pipeline which I leave others to talk about.

Now, we have all the streams from the conference available as Ogg Theora/Video streams, we can also publish them in HTML5 video elements. Check out this Web page which has all the video streams together on a single page. Note that the connections may be a bit dodgy and some drop-outs may occur.

Further, let me recommend the Multimedia Miniconf at, which will take place tomorrow, Tuesday 19th January. The Miniconf has decided to add a talk about “How to stream you conference with open codecs” to help educate any potential future conference organisers and point out the software that helps solve these issues.

UPDATE: I should have stated that I didn’t actually do any of the technical work: it was all done by Ralph Giles, Jan Gerber, and Jan Schmidt.

FOMS and LCA Multimedia Miniconf

If you haven’t proposed a presentation yet, got ahead and register yourself for:

FOMS (Foundations of Open Media Software workshop) at

LCA Multimedia Miniconf at

It’s already November and there’s only Christmas between now and the conferences!

I’m personally hoping for many discussions about HTML5 <video> and <audio>, including what to do with multitrack files, with cue ranges, and captions. These should also be relevant to other open media frameworks – e.g. how should we all handle multitrack sign language tracks?

But there are heaps of other topics to discuss and anyone doing any work with open media software will find a fruitful discussions at FOMS.