Tag Archives: media fragments URI

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.

WebSRT and HTML5 media accessibility

On 23rd July, Ian Hickson, the HTML5 editor, posted an update to the WHATWG mailing list introducing the first draft of a platform for accessibility for the HTML5 <video> element. The platform provides for captions, subtitles, audio descriptions, chapter markers and similar time-synchronized text both in-band (inside the video resource) and out-of-band (as external text files). Right now, the proposal only regards <video>, but I personally believe the same can be applied to the <audio> element, except we have to be a bit more flexible with the rendering approach. Anyway…

What I want to do here is to summarize what was introduced, together with the improvements that I and some others have proposed in follow-up emails, and list some of the media accessibility needs that we are not yet dealing with.

For those wanting to only selectively read some sections, here is a clickable table of contents of this rather long blog post:


The first and to everyone probably most surprising part is the new file format that is being proposed to contain out-of-band time-synchronized text for video. A new format was necessary after the analysis of all relevant existing formats determined that they were either insufficient or hard to use in a Web environment.

The new format is called WebSRT and is an extension to the existing SRT SubRip format. It is actually also the part of the new specification that I am personally most uncomfortable with. Not that WebSRT is a bad format. It’s just not sufficient yet to provide all the functionality that a good time-synchronized text format for Web media should. Let’s look at some examples.

WebSRT is composed of a sequence of timed text cues (that’s what we’ve decided to call the pieces of text that are active during a certain time interval). Because of its ancestry of SRT, the text cues can optionally be numbered through. The content of the text cues is currently allowed to contain three different types of text: plain text, minimal markup, and anything at all (also called “metadata”).

In its most simple form, a WebSRT file is just an ordinary old SRT file with optional cue numbers and only plain text in cues:

  00:00:15.00 --> 00:00:17.95
  At the left we can see...

  00:00:18.16 --> 00:00:20.08
  At the right we can see the...

  00:00:20.11 --> 00:00:21.96
  ...the head-snarlers

A bit of a more complex example results if we introduce minimal markup:

  00:00:15.00 --> 00:00:17.95 A:start
  Auf der <i>linken</i> Seite sehen wir...

  00:00:18.16 --> 00:00:20.08 A:end
  Auf der <b>rechten</b> Seite sehen wir die....

  00:00:20.11 --> 00:00:21.96 A:end
  <1>...die Enthaupter.

  00:00:21.99 --> 00:00:24.36 A:start
  <2>Alles ist sicher.
  Vollkommen <b>sicher</b>.

and add to this a CSS to provide for some colors and special formatting:

    ::cue { background: rgba(0,0,0,0.5); } 
    ::cue-part(1) { color: red; } 
    ::cue-part(2, b) { font-style: normal; text-decoration: underline; } 

Minimal markup accepts <i>, <b>, <ruby> and a timestamp in <>, providing for italics, bold, and ruby markup as well as karaoke timestamps. Any further styling can be done using the CSS pseudo-elements ::cue and ::cue-part, which accept the features ‘color’, ‘text-shadow’, ‘text-outline’, ‘background’, ‘outline’, and ‘font’.

Note that positioning requires some special notes at the end of the start/end timestamps which can provide for vertical text, line position, text position, size and alignment cue setting. Here is an example with vertically rendered Chinese text, right-aligned at 98% of the video frame:

  00:00:15.00 --> 00:00:17.95 A:start D:vertical L:98%

  00:00:18.16 --> 00:00:20.08 A:start D:vertical L:98%

  00:00:20.11 --> 00:00:21.96 A:start D:vertical L:98%

  00:00:21.99 --> 00:00:24.36 A:start D:vertical L:98%

Finally, WebSRT files can be authored with abstract metadata inside cues, which practically means anything at all. Here’s an example with HTML content:

  00:00:15.00 --> 00:00:17.95 A:start
  <img src="pic1.png"/>Auf der <i>linken</i> Seite sehen wir...

  00:00:18.16 --> 00:00:20.08 A:end
  <img src="pic2.png"/>Auf der <b>rechten</b> Seite sehen wir die....

  00:00:20.11 --> 00:00:21.96 A:end
  <img src="pic3.png"/>...die <a href="http://members.chello.nl/j.kassenaar/

  00:00:21.99 --> 00:00:24.36 A:start
  <img src="pic4.png"/>Alles ist <mark>sicher</mark>.<br/>Vollkommen <b>sicher</b>.

Here is another example with JSON in the cues:

  00:00:00.00 --> 00:00:44.00
    slide: intro.png,
    title: "Really Achieving Your Childhood Dreams" by Randy Pausch, 
             Carnegie Mellon University, Sept 18, 2007

  00:00:44.00 --> 00:01:18.00
    slide: elephant.png,
    title: The elephant in the room...

  00:01:18.00 --> 00:02:05.00
    slide: denial.png,
    title: I'm not in denial...

What I like about WebSRT:

  1. it allows for all sorts of different content in the text cues – plain text is useful for texted audio descriptions, minimal markup is useful for subtitles, captions, karaoke and chapters, and “metadata” is useful for, well, any data.
  2. it can be easily encapsulated into media resources and thus turned into in-band tracks by regarding each cue as a data packet with time stamps.
  3. it is not verbose

Where I think WebSRT still needs improvements:

  1. break with the SRT history: since WebSRT and SRT files are so different, WebSRT should get its own MIME type, e.g. text/websrt, and file extensions, e.g. .wsrt; this will free WebSRT for changes that wouldn’t be possible by trying to keep conformant with SRT
  2. introduce some header fields into WebSRT: the format needs
    • file-wide name-value metadata, such as author, date, copyright, etc
    • language specification for the file as a hint for font selection and speech synthesis
    • a possibility for style sheet association in the file header
    • a means to identify which parser is required for the cues
    • a magic identifier and a version string of the format
  3. allow innerHTML as an additional format in the cues with the CSS pseudo-elements applying to all HTML elements
  4. allow full use of CSS instead of just the restricted features and also use it for positioning instead of the hard to understand positioning hints
  5. on the minimum markup, provide a neutral structuring element such as <span @id @class @lang> to associate specific styles or specific languages with a subpart of the cue

Note that I undertook some experiments with an alternative format that is XML-based and called WMML to gain most of these insights and determine the advantages/disadvantages of a xml-based format. The foremost advantage is that there is no automatism with newlines and displayed new lines, which can make the source text file more readable. The foremost disadvantages are verbosity and that there needs to be a simple encoding step to remove all encapsulating header-type content from around the timed text cues before encoding it into a binary media resource.


Now that we have a timed text format, we need to be able to associate it with a media resource in HTML5. This is what the <track> element was introduced for. It associates the timestamps in the timed text cues with the timeline of the video resource. The browser is then expected to render these during the time interval in which the cues are expected to be active.

Here is an example for how to associate multiple subtitle tracks with a video:

  <video src="california.webm" controls>
    <track label="English" kind="subtitles" src="calif_eng.wsrt" srclang="en">
    <track label="German" kind="subtitles" src="calif_de.wsrt" srclang="de">
    <track label="Chinese" kind="subtitles" src="calif_zh.wsrt" srclang="zh">

In this case, the UA is expected to provide a text menu with a subtitle entry with these three tracks and their label as part of the video controls. Thus, the user can interactively activate one of the tracks.

Here is an example for multiple tracks of different kinds:

  <video src="california.webm" controls>
    <track label="English" kind="subtitles" src="calif_eng.wsrt" srclang="en">
    <track label="German" kind="captions" src="calif_de.wsrt" srclang="de">
    <track label="French" kind="chapter" src="calif_fr.wsrt" srclang="fr">
    <track label="English" kind="metadata" src="calif_meta.wsrt" srclang="en">
    <track label="Chinese" kind="descriptions" src="calif_zh.wsrt" srclang="zh">

In this case, the UA is expected to provide a text menu with a list of track kinds with one entry each for subtitles, captions and descriptions through the controls. The chapter tracks are expected to provide some sort of visual subdivision on the timeline and the metadata tracks are not exposed visually, but are only available through the JavaScript API.

Here are several ideas for improving the <track> specification:

  • <track> is currently only defined for WebSRT resources – it should be made generic and then browsers can compete on the formats for which they provide support. WebSRT could be the baseline format. A @type attribute could be added to hint at the MIME type of the provided resource.
  • <track> needs a means for authors to mark certain tracks as active, others as inactive. This can be overruled by browser settings e.g. on @srclang and by user interaction.
  • karaoke and lyrics are supported by WebSRT, but aren’t in the HTML5 spec as track kinds – they should be added and made visible like subtitles or captions.


This is where we take an extra step and move to a uniform handling of both in-band and out-of-band timed text tracks. Futher, a third type of timed text track has been introduced in the form of a MutableTimedTrack – i.e. one that can be authored and added through JavaScript alone.

The JavaScript API that is exposed for any of these track type is identical. A media element now has this additional IDL interface:

interface HTMLMediaElement : HTMLElement {
  readonly attribute TimedTrack[] tracks;
  MutableTimedTrack addTrack(in DOMString label, in DOMString kind, 
                                 in DOMString language);

A media element thus manages a list of TimedTracks and provides for adding TimedTracks through addTrack().

The timed tracks are associated with a media resource in the following order:

  1. The <track> element children of the media element, in tree order.
  2. Tracks created through the addTrack() method, in the order they were added, oldest first.
  3. In-band timed text tracks, in the order defined by the media resource’s format specification.

The IDL interface of a TimedTrack is as follows:

interface TimedTrack {
  readonly attribute DOMString kind;
  readonly attribute DOMString label;
  readonly attribute DOMString language;
  readonly attribute unsigned short readyState;
           attribute unsigned short mode;
  readonly attribute TimedTrackCueList cues;
  readonly attribute TimedTrackCueList activeCues;
  readonly attribute Function onload;
  readonly attribute Function onerror;
  readonly attribute Function oncuechange;

The first three capture the value of the @kind, @label and @srclang attributes and are provided by the addTrack() function for MutableTimedTracks and exposed from metadata in the binary resource for in-band tracks.

The readyState captures whether the data is available and is one of “not loaded”, “loading”, “loaded”, “failed to load”. Data is only availalbe in “loaded” state.

The mode attribute captures whether the data is activate to be displayed and is one of “disabled”, “hidden” and “showing”. In the “disabled” mode, the UA doesn’t have to download the resource, allowing for some bandwidth management.

The cues and activeCues attributes provide the list of parsed cues for the given track and the subpart thereof that is currently active.

The onload, onerror, and oncuechange functions are event handlers for the load, error and cuechange events of the TimedTrack.

Individual cues expose the following IDL interface:

interface TimedTrackCue {
  readonly attribute TimedTrack track;
  readonly attribute DOMString id;
  readonly attribute float startTime;
  readonly attribute float endTime;
  DOMString getCueAsSource();
  DocumentFragment getCueAsHTML();
  readonly attribute boolean pauseOnExit;
  readonly attribute Function onenter;
  readonly attribute Function onexit;
  readonly attribute DOMString direction;
  readonly attribute boolean snapToLines;
  readonly attribute long linePosition;
  readonly attribute long textPosition;
  readonly attribute long size;
  readonly attribute DOMString alignment;
  readonly attribute DOMString voice;

The @track attribute links the cue to its TimedTrack.

The @id, @startTime, @endTime attributes expose a cue identifier and its associated time interval. The getCueAsSource() and getCueAsHTML() functions provide either an unparsed cue text content or a text content parsed into a HTML DOM subtree.

The @pauseOnExit attribute can be set to true/false and indicates whether at the end of the cue’s time interval the media playback should be paused and wait for user interaction to continue. This is particularly important as we are trying to support extended audio descriptions and extended captions.

The onenter and onexit functions are event handlers for the enter and exit events of the TimedTrackCue.

The @direction, @snapToLines, @linePosition, @textPosition, @size, @alignment and @voice attributes expose WebSRT positioning and semantic markup of the cue.

My only concerns with this part of the specification are:

  • The WebSRT-related attributes in the TimedTrackCue are in conflict with CSS attributes and really should not be introduced into HTML5, since they are WebSRT-specific. They will not exist in other types of in-band or out-of-band timed text tracks. As there is a mapping to do already, why not rely on already available CSS features.
  • There is no API to expose header-specific metadata from timed text tracks into JavaScript. This such as the copyright holder, the creation date and the usage rights of a timed text track would be useful to have available. I would propose to add a list of name-value metadata elements to the TimedTrack API.
  • In addition, I would propose to allow media fragment hyperlinks in a <video> @src attribute to point to the @id of a TimedTextCue, thus defining that the playback position should be moved to the time offset of that TimedTextCue. This is a useful feature and builds on bringing named media fragment URIs and TimedTracks together.


The third part of the timed track framework deals with how to render the timed text cues in a Web page. The rendering rules are explained in the HTML5 rendering section.

I’ve extracted the following rough steps from the rendering algorithm:

  1. All timed tracks of a media resource that are in “showing” mode are rendered together to avoid overlapping text from multiple tracks.
  2. The timed tracks cues that are to be rendered are collected from the active timed tracks and ordered by the timed track order first and by their start time second. Where there are identical start times, the cues are ordered by their end time, earliest first, or by their creation order if all else is identical.
  3. Each cue gets its own CSS box.
  4. The text in the CSS boxes is positioned and formated by interpreting the positioning and formatting instructions of WebSRT that are provided on the cues.
  5. An anonymous inline CSS box is created into which all the cue CSS boxes are wrapped.
  6. The wrapping CSS box gets the dimensions of the video viewport. The cue CSS boxes are positioned so they don’t overlap. The text inside the cue CSS boxes inside the wrapping CSS box is wrapped at the edges if necessary.

To overcome security concerns with this kind of direct rendering of a CSS box into the Web page where text comes potentially from a different and malicious Web site, it is required to have the cues come from the same origin as the Web page.

To allow application of a restricted set of CSS properties to the timed text cues, a set of pseudo-selectors was introduced. This is necessary since all the CSS boxes are anonymous and cannot be addressed from the Web page. The introduced pseudo-selectors are ::cue to address a complete cue CSS box, and ::cue-part to address a subpart of a cue CSS box based on a set of identifiers provided by WebSRT.

I have several issues with this approach:

  • I believe that it is not a good idea to only restrict rendering to same-origin files. This will disallow the use of external captioning services (or even just a separate caption server of the same company) to link to for providing the captions to a video. Henri Sivonen proposed a means to overcome this by parsing every cue basically as its own HTML document (well, the body of a document) and then rendering these in iFrame-manner into the Web page. This would overcome the same-origin restriction. It would also allow to do away with the new ::cue CSS selectors, thus simplifying the solution.
  • In general I am concerned about how tightly the rendering is tied to WebSRT. Step 4 should not be in the HTML5 specification, but only apply to WebSRT. Every external format should provide its own mapping to CSS. As it is specified right now, other formats, such as e.g. 3GPP in MPEG-4 or Kate in Ogg, are required to map their format and positioning information to WebSRT instructions. These are then converted again using the WebSRT to CSS mapping rules. That seems overkill.
  • I also find step 6 very limiting, since only the video viewport is regarded as a potential rendering area – this is also the reason why there is no rendering for audio elements. Instead, it would make a lot more sense if a CSS box was provided by the HTML page – the default being the video viewport, but it could be changed to any area on screen. This would allow to render music lyrics under or above an audio element, or render captions below a video element to avoid any overlap at all.


We’ve made huge progress on accessibility features for HTML5 media elements with the specifications that Ian proposed. I think we can move it to a flexible and feature-rich framework as the improvements that Henri, myself and others have proposed are included.

This will meet most of the requirements that the W3C HTML Accessibility Task Force has collected for media elements where the requirements relate to accessibility functionality provided through alternative text resources.

However, we are not solving any of the accessibility needs that relate to alternative audio-visual tracks and resources. In particular there is no solution yet to deal with multi-track audio or video files that have e.g. sign language or audio description tracks in them – not to speak of the issues that can be introduced through dealing with separate media resources from several sites that need to be played back in sync. This latter may be a challenge for future versions of HTML5, since needs for such synchoronisation of multiple resources have to be explored further.

In a first instance, we will require an API to expose in-band tracks, a means to control their activation interactively in a UI, and a description of how they should be rendered. E.g. should a sign language track be rendered as pciture-in-picture? Clear audio and Sign translation are the two key accessibility needs that can be satisfied with such a multi-track solution.

Finally, another key requirement area for media accessibility is described in a section called “Content Navigation by Content Structure”. This describes the need for vision-impaired users to be able to navigate through a media resource based on semantic markup – think of it as similar to a navigation through a book by book chapters and paragraphs. The introduction of chapter markers goes some way towards satisfying this need, but chapter markers tend to address only big time intervals in a video and don’t let you navigate on a different level to subchapters and paragraphs. It is possible to provide that navigation through providing several chapter tracks at different resolution levels, but then they are not linked together and navigation cannot easily swap between resolution levels.

An alternative might be to include different resolution levels inside a single chapter track and somehow control the UI to manage them as different resolutions. This would only require an additional attribute on text cues and could be useful to other types of text tracks, too. For example, captions could be navigated based on scenes, shots, coversations, or individual captions. Some experimentation will be required here before we can introduce a sensible extension to the given media accessibility framework.

URI fragments vs URI queries for media fragment addressing

In the W3C Media Fragment Working Group (MFWG) we have had long discussions about the use of the URI query (“?”) or the URI fragment (“#”) addressing approach for addressing directly into media fragments, and the diverse new HTTP headers required to serve such URI requests, considering such side conditions as the stripping-off of fragment parameters from a URI by Web browsers, or the existence of caching Web proxies.

As explained earlier, URI queries request (primary) resources, while URI fragments address secondary resources, which have a relationship to their primary resource. So, in the strictest sense of their specifications, to address segments in media resources without losing the context of the primary resource, we can only use URI fragments.

Browser-supported Media Fragment URIs

For this reason, URI fragments are also the way in which my last media fragment addressing demo has been implemented. For example, I would address

Media fragment URI addressing

In the media fragment working group at the W3C, we are introducing a standard means to address fragments of media resources through URIs. The idea is to define URIs such as http://example.com/video.ogv#t=24m16s-30m12s, which would only retrieve the subpart of video.ogv that is of interest to the user and thus save bandwidth. This is particularly important for mobile devices, but also for pointing out highlights in videos on the Web, bookmarking, and other use cases.

I’d like to give a brief look into the state of discussion from a technical viewpoint here.

Let’s start by considering the protocols for which such a scheme could be defined. We are currently focusing on HTTP and RTSP, since they are open protocols for media delivery. P2P protocols are also under consideration, however, most of them are proprietary. Also, p2p protocols are mostly used to transfer complete large files, so fragment addressing may not be desired. RSTP already has a mechanism to address temporal fragments of media resources through a range parameter of the play request as part of the protocol parameters. Yet, there is no URI addressing scheme for this. Our key focus however is HTTP, since most video content nowadays is transferred over HTTP, e.g. YouTube.

Another topic that needs discussion are the types of fragmentation for which we will specify addressing schemes. At the moment, we are considering temporal fragmentation, spatial fragmentation, and fragmentation by tracks. In temporal fragmentation, a request asks for a time interval that is a subpart of the media resource (e.g. audio or video). In spatial fragmentation, the request is for an image region (e.g. in an image or a video). Track fragmentation addresses the issue where, e.g. a blind person would not require to receive the actual video data for a video and thus a user agent could request only the data tracks from the resource that are really required for the user.

Another concern is the syntax of URI addressing. URI fragments (“#”) have been invented to created URIs that point at so-called “secondary” resources. Per definition, a secondary resource may be some portion or subset of the primary resource, some view on representations of the primary resource, or some other resource defined or described by those representations. It is therefore the perfect syntax for media fragment URIs.

The only issue is that URI fragments (“#”) are not expected to be transferred from the client to the server (e.g. Apache strips it off the URI if it receives it). Therefore, in the temporal URI specification of Annodex we decided to use the query (“?”) parameter instead. This is however not necessary. The W3C working group is proposing to have the user agent strip off the URI fragment specification and transform it into a protocol parameter. For HTTP, the idea is to introduce new range units for the types of fragmentation that we will define. Then, the Range and Content-Range headers can be used to request and deliver the information about the fragmentation.

The most complicated issue that we are dealing with is the issue of caching in Web proxies. Existing Web proxies will not be able to understand new range units and will therefore not cache such requests. This is unfortunate and we are trying to devise two schemes – one for existing Web proxies and one for future more intelligent Web proxies – to enable proxy caching. This discussion has many dimensions – such as e.g. the ability to uniquely map time to bytes for any codec format, the ability to recompose new fragment requests from existing combined fragment requests, or the need and abilities for partial re-encoding. Mostly we are dealing with the complexities and restrictions of different codecs and encapsulation formats. Possibly, the idea of recomposition of ranges in Web proxies is too complex to realise and caching is best done by regarding each fragment as its own cacheable resource, but this hasn’t been decided yet.

We now have experts from the squid community, from YouTube/Google, HTTP experts, Web accessibility experts, SMIL experts, me from Annodex/Xiph, and a more people with diverse media backgrounds in the team. It’s a great group and we are covering the issues from all aspects. The brief update above is given from my perspective, and only lists the key issues superficially, while the discussions that we’re having on the mailing list and in meetings are much more in-depth.

I am not quite expecting us to meet the deadline of having a first working draft before the end of this month, but certainly before Christmas.