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• FFmpeg "Non-monotonous DTS in output stream" error when processing video from Safari's MediaRecorder

  17 juillet 2024, par Hackermon

  I'm recording a video stream in Safari with MediaRecorder, then sending it to a remote server which then uses ffmpeg to reencode the video. When reencoding with FFmpeg, I get a lot of warnings and the final video is broken, frame are glitching and out of sync but the audio sounds fine.

  


  Here's my MediaRecorder script :

  


  const camera = await navigator.mediaDevices.getUserMedia({ audio: true, video: true });
const recorder = new MediaRecorder(camera, {
        mimeType: 'video/mp4', // Safari only supports MP4
        bitsPerSecond: 1_000_000,
});

recorder.ondataavailable = async ({ data: blob }) => {
        // open contents in new tab
        var fileURL = URL.createObjectURL(file);
         window.open(fileURL, '_blank');
};

recorder.start();
setTimeout(() => recorder.stop(), 5000);


  


  I download the video blob from Safari and use this command to reencode it :

  


  ffmpeg -i ./blob.mp4 -preset ultrafast -strict -2 -threads 10 -c copy ./output.mp4


  


  Logs :

  


  ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
  configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x559826616000] DTS 29 < 313 out of order
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './chunk1.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2024-07-17T14:30:47.000000Z
  Duration: 00:00:01.00, start: 0.000000, bitrate: 3937 kb/s
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], 6218 kb/s, 33.36 fps, 600 tbr, 600 tbn, 1200 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
File './safari3.mp4' already exists. Overwrite ? [y/N] Output #0, mp4, to './safari3.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], q=2-31, 6218 kb/s, 33.36 fps, 600 tbr, 19200 tbn, 600 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10016, current: 928; changing to 10017. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10017, current: 1568; changing to 10018. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10018, current: 2208; changing to 10019. This may result in incorrect timestamps in the output file.
...x100
frame=  126 fps=0.0 q=-1.0 Lsize=     479kB time=00:00:00.97 bitrate=4026.2kbits/s speed= 130x    
video:425kB audio:51kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.652696%


  


  Not sure what's happening or how to fix it. This issue only happens in Safari, videos from Chrome are perfectly fine.

  


  I've tried various flags :

  


  -fflags +igndts
-bsf:a aac_adtstoasc
-c:v libvpx-vp9 -c:a libopus
etc


  


  None of them seem to fix the issue.

  


• xdotool to tab to a button on a web page and use the mouse to disable a drop down menu option

  25 juin 2023, par Mash

  I have a Bash script that open a Amazon chime meeting URL in firefox, uses XDOtool to enter a meetig participant name, and tab and mouse click functions. and next uses ffmpeg to stream the video and audio output of the Amazon chime meeting to an RTMP destination.

  


  When this is streamed, the Amazon chime web app has "More" drop down Menu. Within the Menu it has a option to disable the self view. I want to add xdotool commands to disable this self view option from the more drop down menu on the amazon chime web app page.

  


  the Amazon chime meeting URL is - https://app.chime.aws/meetings/

  


  Here is the Bash Script

  


  #!/bin/bash&#xA;BROWSER_URL=${MEETING_URL}&#xA;SCREEN_WIDTH=1920&#xA;SCREEN_HEIGHT=1080&#xA;SCREEN_RESOLUTION=${SCREEN_WIDTH}x${SCREEN_HEIGHT}&#xA;CAPTURE_SCREEN_RESOLUTION=1920x1080&#xA;COLOR_DEPTH=24&#xA;X_SERVER_NUM=2&#xA;VIDEO_BITRATE=6000&#xA;VIDEO_FRAMERATE=30&#xA;VIDEO_GOP=$((VIDEO_FRAMERATE * 2))&#xA;AUDIO_BITRATE=160k&#xA;AUDIO_SAMPLERATE=44100&#xA;AUDIO_CHANNELS=2&#xA;&#xA;# Start PulseAudio server so Firefox will have somewhere to which to send audio&#xA;pulseaudio -D --exit-idle-time=-1&#xA;pacmd load-module module-virtual-sink sink_name=v1  # Load a virtual sink as `v1`&#xA;pacmd set-default-sink v1  # Set the `v1` as the default sink device&#xA;pacmd set-default-source v1.monitor  # Set the monitor of the v1 sink to be the default source&#xA;&#xA;# Start X11 virtual framebuffer so Firefox will have somewhere to draw&#xA;Xvfb :${X_SERVER_NUM} -ac -screen 0 ${SCREEN_RESOLUTION}x${COLOR_DEPTH} > /dev/null 2>&amp;1 &amp;&#xA;export DISPLAY=:${X_SERVER_NUM}.0&#xA;sleep 0.5  # Ensure this has started before moving on&#xA;&#xA;# Create a new Firefox profile for capturing preferences for this&#xA;firefox --no-remote --new-instance --createprofile "foo4 /tmp/foo4"&#xA;&#xA;# Install the OpenH264 plugin for Firefox&#xA;mkdir -p /tmp/foo4/gmp-gmpopenh264/1.8.1.1/&#xA;pushd /tmp/foo4/gmp-gmpopenh264/1.8.1.1 >&amp; /dev/null&#xA;curl -s -O http://ciscobinary.openh264.org/openh264-linux64-2e1774ab6dc6c43debb0b5b628bdf122a391d521.zip&#xA;unzip openh264-linux64-2e1774ab6dc6c43debb0b5b628bdf122a391d521.zip&#xA;rm -f openh264-linux64-2e1774ab6dc6c43debb0b5b628bdf122a391d521.zip&#xA;popd >&amp; /dev/null&#xA;&#xA;# Set the Firefox preferences to enable automatic media playing with no user&#xA;# interaction and the use of the OpenH264 plugin.&#xA;cat &lt;<eof>> /tmp/foo4/prefs.js&#xA;user_pref("media.autoplay.default", 0);&#xA;user_pref("media.autoplay.enabled.user-gestures-needed", false);&#xA;user_pref("media.navigator.permission.disabled", true);&#xA;user_pref("media.gmp-gmpopenh264.abi", "x86_64-gcc3");&#xA;user_pref("media.gmp-gmpopenh264.lastUpdate", 1571534329);&#xA;user_pref("media.gmp-gmpopenh264.version", "1.8.1.1");&#xA;user_pref("doh-rollout.doorhanger-shown", true);&#xA;EOF&#xA;&#xA;# Start Firefox browser and point it at the URL we want to capture&#xA;#&#xA;# NB: The `--width` and `--height` arguments have to be very early in the&#xA;# argument list or else only a white screen will result in the capture for some&#xA;# reason.&#xA;firefox \&#xA;  -P foo4 \&#xA;  --width ${SCREEN_WIDTH} \&#xA;  --height ${SCREEN_HEIGHT} \&#xA;  --new-instance \&#xA;  --first-startup \&#xA;  --foreground \&#xA;  --kiosk \&#xA;  --ssb \&#xA;  "${BROWSER_URL}" \&#xA;  &amp;&#xA;sleep 10  # Ensure this has started before moving on, waiting for loading the Chime web app&#xA;xdotool key Return #Select yes for the pop-up window of "Would you like to open this link with Chime app?"&#xA;sleep 3&#xA;xdotool key Escape #Close the pop-up window&#xA;sleep 3&#xA;xdotool type Livestream #Type "Livestream" on the name input field&#xA;sleep 3&#xA;xdotool key Tab #Move to "join the meeting" button&#xA;sleep 3&#xA;xdotool key Return #Click "join the meeting" button&#xA;sleep 3&#xA;xdotool key Return #Close the pop-up window once again&#xA;sleep 3&#xA;xdotool key Escape #Close the pop-up window once again&#xA;sleep 3&#xA;xdotool key Return #Click "Use system audio" setting&#xA;sleep 3&#xA;xdotool key Escape #Close warning message&#xA;sleep 3&#xA;xdotool mousemove 1 1 click 1  # Move mouse out of the way so it doesn&#x27;t trigger the "pause" overlay on the video tile  &#xA;&#xA;# Start ffmpeg to transcode the capture from the X11 framebuffer and the&#xA;# PulseAudio virtual sound device we created earlier and send that to the RTMP&#xA;# endpoint in H.264/AAC format using a FLV container format.&#xA;#&#xA;# NB: These arguments have a very specific order. Seemingly inocuous changes in&#xA;# argument order can have pretty drastic effects, so be careful when&#xA;# adding/removing/reordering arguments here.&#xA;ffmpeg \&#xA;  -hide_banner -loglevel error \&#xA;  -nostdin \&#xA;  -s ${CAPTURE_SCREEN_RESOLUTION} \&#xA;  -r ${VIDEO_FRAMERATE} \&#xA;  -draw_mouse 0 \&#xA;  -f x11grab \&#xA;    -i ${DISPLAY} \&#xA;  -f pulse \&#xA;    -ac 2 \&#xA;    -i default \&#xA;    -vf "crop=1600:980:0:1080" \&#xA;  -c:v libx264 \&#xA;    -pix_fmt yuv420p \&#xA;    -profile:v main \&#xA;    -preset slow \&#xA;    -x264opts "nal-hrd=cbr:no-scenecut" \&#xA;    -minrate ${VIDEO_BITRATE} \&#xA;    -maxrate ${VIDEO_BITRATE} \&#xA;    -g ${VIDEO_GOP} \&#xA;  -filter_complex "aresample=async=1000:min_hard_comp=0.100000:first_pts=1" \&#xA;  -async 1 \&#xA;  -c:a aac \&#xA;    -b:a ${AUDIO_BITRATE} \&#xA;    -ac ${AUDIO_CHANNELS} \&#xA;    -ar ${AUDIO_SAMPLERATE} \&#xA;  -f flv ${RTMP_URL}``&#xA;&#xA;</eof>

  


  what i have tried so far in in the bash script

  


• 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.