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• Exceeded GA’s 10M hits data limit, now what ?

  21 juin 2019, par Joselyn Khor
  Exceeded GA’s 10M hits data limit, now what ? Matomo has the answers

  “Your data volume (1XXM hits) exceeds the limit of 10M hits per month as outlined in our Terms of Service. If you continue to exceed the limit, we will stop processing new data on XXX 21, 2019. Learn more about possible solutions.”

  Yikes. Alarm bells were ringing when a Google Analytics free user came to us faced with this notice. Let’s call him ‘Mark’. Mark had reached the limits on the data he could collect through Google Analytics and was shocked by the limited options available to fix the problem, without blowing the budget. The thoughts racing through his head were :

  • “What happens to all my data ?”
  • “What if Google starts charging USD150K now ?”

  Then he came across Matomo and decided to get in touch with our support team …

  “Can you fix this issue ?” he asked us.

  “Absolutely !” we said.

  We’ll get back to helping Mark in a minute. For now let’s go over why this was such a dilemma for him.

  In order to resolve this data limits issue, one of the solutions was for him to upgrade to Google Analytics 360, which meant shelling out USD150,000 per year for their 1 billion hits per month option. Going from free to USD150,000 was too much of a stretch for a growing company.

  “Your data volume (1XXM hits) exceeds the limit of 10M hits per month …”, what did this message mean ?

  With the free version, Mark could collect up to 10 million “hits” per month, per account. Going over meant Google Analytics could stop collecting any more data for free as outlined in their Terms.

  Google Analytics’ Terms of Service (2018, sec. 2) states, “Subject to Section 15, the Service is provided without charge to You for up to 10 million Hits per month per account.”[1]

  In general, what’s a "hit" ?

  Data being sent to Google Analytics. It can be a transaction, event, social interaction or pageview - these all produce what Google calls a “hit”.

  

  And their Analytics Help Data Limits (n.d.) support page makes clear that : “If a property sends more hits per month to Analytics than allowed by the Analytics Terms of Service, there is no assurance that the excess hits will be processed. If the property’s hit volume exceeds this limit, a warning may be displayed in the user interface and you may be prevented from accessing reports.”[2]

  

  Possible solutions

  So the possible solutions given by Google Analytics’ Data Limits support page were (also shown in image below) :

  • To pay USD150K to upgrade to Google Analytics 360
  • To send fewer hits by setting up sampling
  • Or choose the slightly less relevant option to upgrade mobile app tracking to Google Analytics for Firebase.

  Without the means to pay, the free version was fast becoming inaccessible for Mark as he was facing a future where he risked no longer having access to up-to-date data used in his business’ reporting.

  Mark was facing a problem that potentially didn’t have a cost-effective solution.

  

  So what can you really do about it ?

  This is where we can help provide some assistance. If you’re reading this article, we’ll assume you can relate to Mark and share with you the advice on options we gave him.

  Options :

  One option posed by Google is for you to send fewer hits by auditing your data collection processes

  If you really don’t have the budget, you’ll need to reassess your data collection priorities and go over your strategies to see what is necessary to track, and what isn’t.

  • Make sure you know what you’re tracking and why. Look at what websites are being tracked by Google and into what properties.
  • Go through what data you’re tracking and decide what is or isn’t of value.
  • Set up data sampling, this however, will lead to inaccurate data.

  From here you can start to course correct. If you’ve found data you’re not using for analysis, get rid of these events/pageviews in your Google Analytics.

  But the limitations here are that eventually, you’re going to run out of irrelevant metrics and everything you’re tracking will be essential. So you’ll hit another brick wall and return to the same situation.

  Option 2 – Ignore and continue using the free version of Google Analytics

  With this option, you’ll have to bear the business risks involved by basing decisions off of analytics reports that may or may not be updated. In this case, you may still get contacted about exceeding the limits. As the free service is provided for only up to 10 million hits, once you’ve gone over them, you’re violating what’s stipulated in the Terms of Service. 

  There’s also the warning that “… you may be prevented from accessing reports” (Data limits, n.d.). So while we may not know for certain what Google Analytics will do, in this case it may be better to be safe rather than sorry by acting quickly to resolve it. 

  Option 3 –The Matomo solution. Upgrade to a web analytics platform that can handle your demanding data requirements

  Save money while continuing to gain valuable insights by moving over to Matomo Analytics (recommended)

  This is where you can save up to USD130,000 a year. As well as that, the transition from Google Analytics to the Matomo Cloud is a seamless experience as setup and maintenance is taken care of by our experts.

  For example, you can get up to 15M pageviews for USD1,612.50/month (or USD19,350/year) on the Essentials plan.

  Or even 25M pageviews for USD2400/month (or USD24000/year) on the Business plan – which offers additional web analytics and conversion optimization resources.

  Matomo Cloud is a great option if you’re looking for a secure, cost-effective and powerful analytics solution. You also get what Google Analytics could never offer you : full control and ownership of your own data and privacy. 

  No need to worry about losing your Google Analytics data because …

  Now you can import your historic Google Analytics data directly into your Matomo with the Google Analytics Importer tool. Simply follow the step-by-step guide to get started for free.

  Along with savings you can get :

  • A solution for the data limits issue forever. You choose the right plan to suit your data needs and adapt as you continue growing
  • 100% accurate data (no data sampling)
  • 100% data ownership of all your information without signing away your data to a third party
  • Powerful web analytics and conversion optimization features
  • Matomo Tag Manager
  • Easy setup
  • Support from Matomo’s specialists

  Learn more about Matomo Cloud pricing.

  

  Or go for Matomo On-Premise

  If you have the in-house infrastructure to support self-hosting Matomo on your own servers then there’s also the option of Matomo On-Premise. Here you’ll get full security knowing the data is on your own servers. 

  Setup will also require technical knowledge. There will also be costs associated with acquiring your own servers, and keeping up with regular maintenance and updates. With On-Premise you get maximum flexibility, with no data limits whatsoever. But if you’re coming over from Google Analytics and don’t have the infrastructure and team to host On-Premise, the Matomo Cloud could be right for you.

  Learn more about Matomo On-Premise.

  Where do you go from here ?

  Getting 10 millions hits per month is no small feat, it’s actually pretty fantastic. But if it means having to shell out USD150,000 just to be able to continue with Google Analytics, we feel your problem could be fixed with Matomo Cloud. You could then put the rest of the money you save to better use.

  If you choose Matomo, you now have the option to : 

  • Raise your data limits for a fraction of Google Analytics 360’s price
  • Get a comprehensive range of analytics features for the most impactful insights to ensure your website continues excelling
  • Get data that’s not sampled – meaning 100% accuracy in your reports
  • Migrate your data easily with the help of Matomo’s support team

  We’ll have you covered. 

  By sharing with you the options and advice we gave to Mark, we hope you’ll be able to find a solution that makes your life easier and solves the issue of data restrictions forever.

  The team at Matomo is here to help you every step of the way to ensure a stress-free transition from Google Analytics if that is what works best for you.

  For next steps, why not check out our pricing page to see what could suit your needs !

  References :

  [1] Terms of Service. (2018, July 24). In Google Analytics Terms of Service. Retrieved June 12, 2019, from https://www.google.com/analytics/terms/us.html

  [2] Data limits. (n.d.). In Analytics Help Data limits. Retrieved June 12, 2019, from https://support.google.com/analytics/answer/1070983?hl=en

• Open Media Developers Track at OVC 2011

  11 octobre 2011, par silvia

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

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

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

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

  WebRTC : Realtime Communications and HTML5

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

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

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

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

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

  The Missing Link : Flash to HTML5

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

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

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

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

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

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

  Standards for Video Accessibility

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

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

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

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

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

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

  WebM : Testing, Metrics and New features

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

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

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

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

  Standards for HTTP Adaptive Streaming

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

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

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

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

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

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

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

  The proposed extension to the HTMLMediaElement is as follows :

  partial interface HTMLMediaElement 
    // URL passed to src attribute to enable the media source logic.
    readonly attribute [URL] DOMString webkitMediaSourceURL ;
    bool webkitSourceAppend(in Uint8Array data) ;
    // end of stream status codes.
  
    const unsigned short EOS_NO_ERROR = 0 ;
  
    const unsigned short EOS_NETWORK_ERR = 1 ;
  
    const unsigned short EOS_DECODE_ERR = 2 ;
    void webkitSourceEndOfStream(in unsigned short status) ;
    // states
  
    const unsigned short SOURCE_CLOSED = 0 ;
  
    const unsigned short SOURCE_OPEN = 1 ;
  
    const unsigned short SOURCE_ENDED = 2 ;
    readonly attribute unsigned short webkitSourceState ;
  
   ;
  

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

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

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

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

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

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

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

  Standards for Browser Video Playback Metrics

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

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

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

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

  Device Inputs / A/V in the Browser

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

  Open Video Editors

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

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

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

  Monday’s Breakout Groups

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

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

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

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

  Thanks go to Google for sponsoring my trip to OVC.

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

• Ogg objections

  3 mars 2010, par Mans — Multimedia

  The Ogg container format is being promoted by the Xiph Foundation for use with its Vorbis and Theora codecs. Unfortunately, a number of technical shortcomings in the format render it ill-suited to most, if not all, use cases. This article examines the most severe of these flaws.
  

  Overview of Ogg

  The basic unit in an Ogg stream is the page consisting of a header followed by one or more packets from a single elementary stream. A page can contain up to 255 packets, and a packet can span any number of pages. The following table describes the page header.

  Field	Size (bits)	Description
  capture_pattern	32	magic number “OggS”
  version	8	always zero
  flags	8
  granule_position	64	abstract timestamp
  bitstream_serial_number	32	elementary stream number
  page_sequence_number	32	incremented by 1 each page
  checksum	32	CRC of entire page
  page_segments	8	length of segment_table
  segment_table	variable	list of packet sizes

  Elementary stream types are identified by looking at the payload of the first few pages, which contain any setup data required by the decoders. For full details, see the official format specification.

  Generality

  Ogg, legend tells, was designed to be a general-purpose container format. To most multimedia developers, a general-purpose format is one in which encoded data of any type can be encapsulated with a minimum of effort.

  The Ogg format defined by the specification does not fit this description. For every format one wishes to use with Ogg, a complex mapping must first be defined. This mapping defines how to identify a codec, how to extract setup data, and even how timestamps are to be interpreted. All this is done differently for every codec. To correctly parse an Ogg stream, every such mapping ever defined must be known.

  Under this premise, a centralised repository of codec mappings would seem like a sensible idea, but alas, no such thing exists. It is simply impossible to obtain a exhaustive list of defined mappings, which makes the task of creating a complete implementation somewhat daunting.

  One brave soul, Tobias Waldvogel, created a mapping, OGM, capable of storing any Microsoft AVI compatible codec data in Ogg files. This format saw some use in the wild, but was frowned upon by Xiph, and it was eventually displaced by other formats.

  True generality is evidently not to be found with the Ogg format.

  A good example of a general-purpose format is Matroska. This container can trivially accommodate any codec, all it requires is a unique string to identify the codec. For codecs requiring setup data, a standard location for this is provided in the container. Furthermore, an official list of codec identifiers is maintained, meaning all information required to fully support Matroska files is available from one place.

  Matroska also has probably the greatest advantage of all : it is in active, wide-spread use. Historically, standards derived from existing practice have proven more successful than those created by a design committee.

  Overhead

  When designing a container format, one important consideration is that of overhead, i.e. the extra space required in addition to the elementary stream data being combined. For any given container, the overhead can be divided into a fixed part, independent of the total file size, and a variable part growing with increasing file size. The fixed overhead is not of much concern, its relative contribution being negligible for typical file sizes.

  The variable overhead in the Ogg format comes from the page headers, mostly from the segment_table field. This field uses a most peculiar encoding, somewhat reminiscent of Roman numerals. In Roman times, numbers were written as a sequence of symbols, each representing a value, the combined value being the sum of the constituent values.

  The segment_table field lists the sizes of all packets in the page. Each value in the list is coded as a number of bytes equal to 255 followed by a final byte with a smaller value. The packet size is simply the sum of all these bytes. Any strictly additive encoding, such as this, has the distinct drawback of coded length being linearly proportional to the encoded value. A value of 5000, a reasonable packet size for video of moderate bitrate, requires no less than 20 bytes to encode.

  On top of this we have the 27-byte page header which, although paling in comparison to the packet size encoding, is still much larger than necessary. Starting at the top of the list :

  • The version field could be disposed of, a single-bit marker being adequate to separate this first version from hypothetical future versions. One of the unused positions in the flags field could be used for this purpose
  • A 64-bit granule_position is completely overkill. 32 bits would be more than enough for the vast majority of use cases. In extreme cases, a one-bit flag could be used to signal an extended timestamp field.
  • 32-bit elementary stream number ? Are they anticipating files with four billion elementary streams ? An eight-bit field, if not smaller, would seem more appropriate here.
  • The 32-bit page_sequence_number is inexplicable. The intent is to allow detection of page loss due to transmission errors. ISO MPEG-TS uses a 4-bit counter per 188-byte packet for this purpose, and that format is used where packet loss actually happens, unlike any use of Ogg to date.
  • A mandatory 32-bit checksum is nothing but a waste of space when using a reliable storage/transmission medium. Again, a flag could be used to signal the presence of an optional checksum field.

  With the changes suggested above, the page header would shrink from 27 bytes to 12 bytes in size.

  We thus see that in an Ogg file, the packet size fields alone contribute an overhead of 1/255 or approximately 0.4%. This is a hard lower bound on the overhead, not attainable even in theory. In reality the overhead tends to be closer to 1%.

  Contrast this with the ISO MP4 file format, which can easily achieve an overhead of less than 0.05% with a 1 Mbps elementary stream.

  Latency

  In many applications end-to-end latency is an important factor. Examples include video conferencing, telephony, live sports events, interactive gaming, etc. With the codec layer contributing as little as 10 milliseconds of latency, the amount imposed by the container becomes an important factor.

  Latency in an Ogg-based system is introduced at both the sender and the receiver. Since the page header depends on the entire contents of the page (packet sizes and checksum), a full page of packets must be buffered by the sender before a single bit can be transmitted. This sets a lower bound for the sending latency at the duration of a page.

  On the receiving side, playback cannot commence until packets from all elementary streams are available. Hence, with two streams (audio and video) interleaved at the page level, playback is delayed by at least one page duration (two if checksums are verified).

  Taking both send and receive latencies into account, the minimum end-to-end latency for Ogg is thus twice the duration of a page, triple if strict checksum verification is required. If page durations are variable, the maximum value must be used in order to avoid buffer underflows.

  Minimum latency is clearly achieved by minimising the page duration, which in turn implies sending only one packet per page. This is where the size of the page header becomes important. The header for a single-packet page is 27 + packet_size/255 bytes in size. For a 1 Mbps video stream at 25 fps this gives an overhead of approximately 1%. With a typical audio packet size of 400 bytes, the overhead becomes a staggering 7%. The average overhead for a multiplex of these two streams is 1.4%.

  As it stands, the Ogg format is clearly not a good choice for a low-latency application. The key to low latency is small packets and fine-grained interleaving of streams, and although Ogg can provide both of these, by sending a single packet per page, the price in overhead is simply too high.

  ISO MPEG-PS has an overhead of 9 bytes on most packets (a 5-byte timestamp is added a few times per second), and Microsoft’s ASF has a 12-byte packet header. My suggestions for compacting the Ogg page header would bring it in line with these formats.

  Random access

  Any general-purpose container format needs to allow random access for direct seeking to any given position in the file. Despite this goal being explicitly mentioned in the Ogg specification, the format only allows the most crude of random access methods.

  While many container formats include an index allowing a time to be directly translated into an offset into the file, Ogg has nothing of this kind, the stated rationale for the omission being that this would require a two-pass multiplexing, the second pass creating the index. This is obviously not true ; the index could simply be written at the end of the file. Those objecting that this index would be unavailable in a streaming scenario are forgetting that seeking is impossible there regardless.

  The method for seeking suggested by the Ogg documentation is to perform a binary search on the file, after each file-level seek operation scanning for a page header, extracting the timestamp, and comparing it to the desired position. When the elementary stream encoding allows only certain packets as random access points (video key frames), a second search will have to be performed to locate the entry point closest to the desired time. In a large file (sizes upwards of 10 GB are common), 50 seeks might be required to find the correct position.

  A typical hard drive has an average seek time of roughly 10 ms, giving a total time for the seek operation of around 500 ms, an annoyingly long time. On a slow medium, such as an optical disc or files served over a network, the times are orders of magnitude longer.

  A factor further complicating the seeking process is the possibility of header emulation within the elementary stream data. To safeguard against this, one has to read the entire page and verify the checksum. If the storage medium cannot provide data much faster than during normal playback, this provides yet another substantial delay towards finishing the seeking operation. This too applies to both network delivery and optical discs.

  Although optical disc usage is perhaps in decline today, one should bear in mind that the Ogg format was designed at a time when CDs and DVDs were rapidly gaining ground, and network-based storage is most certainly on the rise.

  The final nail in the coffin of seeking is the codec-dependent timestamp format. At each step in the seeking process, the timestamp parsing specified by the codec mapping corresponding the current page must be invoked. If the mapping is not known, the best one can do is skip pages until one with a known mapping is found. This delays the seeking and complicates the implementation, both bad things.

  Timestamps

  A problem old as multimedia itself is that of synchronising multiple elementary streams (e.g. audio and video) during playback ; badly synchronised A/V is highly unpleasant to view. By the time Ogg was invented, solutions to this problem were long since explored and well-understood. The key to proper synchronisation lies in tagging elementary stream packets with timestamps, packets carrying the same timestamp intended for simultaneous presentation. The concept is as simple as it seems, so it is astonishing to see the amount of complexity with which the Ogg designers managed to imbue it. So bizarre is it, that I have devoted an entire article to the topic, and will not cover it further here.

  Complexity

  Video and audio decoding are time-consuming tasks, so containers should be designed to minimise extra processing required. With the data volumes involved, even an act as simple as copying a packet of compressed data can have a significant impact. Once again, however, Ogg lets us down. Despite the brevity of the specification, the format is remarkably complicated to parse properly.

  The unusual and inefficient encoding of the packet sizes limits the page size to somewhat less than 64 kB. To still allow individual packets larger than this limit, it was decided to allow packets spanning multiple pages, a decision with unfortunate implications. A page-spanning packet as it arrives in the Ogg stream will be discontiguous in memory, a situation most decoders are unable to handle, and reassembly, i.e. copying, is required.

  The knowledgeable reader may at this point remark that the MPEG-TS format also splits packets into pieces requiring reassembly before decoding. There is, however, a significant difference there. MPEG-TS was designed for hardware demultiplexing feeding directly into hardware decoders. In such an implementation the fragmentation is not a problem. Rather, the fine-grained interleaving is a feature allowing smaller on-chip buffers.

  Buffering is also an area in which Ogg suffers. To keep the overhead down, pages must be made as large as practically possible, and page size translates directly into demultiplexer buffer size. Playback of a file with two elementary streams thus requires 128 kB of buffer space. On a modern PC this is perhaps nothing to be concerned about, but in a small embedded system, e.g. a portable media player, it can be relevant.

  In addition to the above, a number of other issues, some of them minor, others more severe, make Ogg processing a painful experience. A selection follows :

  • 32-bit random elementary stream identifiers mean a simple table-lookup cannot be used. Instead the list of streams must be searched for a match. While trivial to do in software, it is still annoying, and a hardware demultiplexer would be significantly more complicated than with a smaller identifier.
  • Semantically ambiguous streams are possible. For example, the continuation flag (bit 1) may conflict with continuation (or lack thereof) implied by the segment table on the preceding page. Such invalid files have been spotted in the wild.
  • Concatenating independent Ogg streams forms a valid stream. While finding a use case for this strange feature is difficult, an implementation must of course be prepared to encounter such streams. Detecting and dealing with these adds pointless complexity.
  • Unusual terminology : inventing new terms for well-known concepts is confusing for the developer trying to understand the format in relation to others. A few examples :
    

    Ogg name	Usual name
    logical bitstream	elementary stream
    grouping	multiplexing
    lacing value	packet size (approximately)
    segment	imaginary element serving no real purpose
    granule position	timestamp

  Final words

  We have found the Ogg format to be a dubious choice in just about every situation. Why then do certain organisations and individuals persist in promoting it with such ferocity ?

  When challenged, three types of reaction are characteristic of the Ogg campaigners.

  On occasion, these people will assume an apologetic tone, explaining how Ogg was only ever designed for simple audio-only streams (ignoring it is as bad for these as for anything), and this is no doubt true. Why then, I ask again, do they continue to tout Ogg as the one-size-fits-all solution they already admitted it is not ?

  More commonly, the Ogg proponents will respond with hand-waving arguments best summarised as Ogg isn’t bad, it’s just different. My reply to this assertion is twofold :

  • Being too different is bad. We live in a world where multimedia files come in many varieties, and a decent media player will need to handle the majority of them. Fortunately, most multimedia file formats share some basic traits, and they can easily be processed in the same general framework, the specifics being taken care of at the input stage. A format deviating too far from the standard model becomes problematic.
  • Ogg is bad. When every angle of examination reveals serious flaws, bad is the only fitting description.

  The third reaction bypasses all technical analysis : Ogg is patent-free, a claim I am not qualified to directly discuss. Assuming it is true, it still does not alter the fact that Ogg is a bad format. Being free from patents does not magically make Ogg a good choice as file format. If all the standard formats are indeed covered by patents, the only proper solution is to design a new, good format which is not, this time hopefully avoiding the old mistakes.