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jbk
BlackLotus89
A bit off topic, but in your talk you said that intel, nvidia and amd encoder are worse than svt-av1. How much worse and which card is the best? Always encoding with ab-av1/av1an using svt-av1, but with the "right" preset it's still really slow. Which hardware encoder (in your opinion/experience) is the best to use (size/quality speed doesn't matter as much). If I want to convert my media server to av1 only should I stick with svt-av1 and eat the power/time-cost?
cstejerean
What is the the "right" preset? SVT-AV1 has a wide range of presets, from being about as fast as x264 veryfast onwards. And for any given speed level it should provide significantly lower bitrates at the same quality than x264, x265 or libvpx.
For example if you pick speed 6 on SVT-AV1 it should take about as much time as x264 veryslow while still being 30-40% lower bitrate at the same quality.
Have you tried the faster presets of SVT-AV1?
brucethemoose2
I dont have a card to test, but the last VMAF bench I saw shows hardware AV1 is slightly better than veryslow x264.
Which is way better than hardware h.264... but also terrible compared to svt-av1, lol.
Try using av1an to parallelize (and improve the quality of) your encode. And TBH you should not re encode your library unless it has raw rips or something like that, and you are short on space.
adverbly
> 200k LoC of handwritten ASM!
He's not the hero we deserve, but he's the hero we need.
I'm sure you get this a lot, but thank you so much for your sacrifice!
AndyKelley
Thank you for doing this talk. I agree with you, news about ffmpeg progress should be more widespread. And by the way, I love your hat!
MezzoDelCammin
nice talk, thanks! I guess for once You finally "did" make it with FFmpeg to the front page of HN ;-)
jbk
Thanks for watching the talk :)
mikae1
At 2m15s you even mention HN. :)
qwerty456127
This is very nice. I've been surprised how many people don't know HN, it can definitely use some promotion.
eigart
Be careful what you wish for!
_0ffh
Please, no!
We'd need way more dangs than are available.
WithinReason
Rule 1 of Fight Club
ikt
no ty!
SuperSandro2000
git log is no changelog...
vhiremath4
Gracias!
stefantalpalaru
[dead]
lxe
Fun fact of the day: ffmpeg is written by Fabrice Bellard, who among other impressive things, wrote the JS PC emulator capable of running windows 2000 in a browser over 10 years ago that got me fascinated with emulators in general.
Jasper_
Fabrice has not been involved in ffmpeg for well over 15 years now.
sirsinsalot
Fabrice is superhuman for sure, but in 20 years the urban legend will go like ...
Fabrice Bellard once read all of TAOCP in 20 minutes before being absorbed into his own AI and is now ascended in the ethereal ethernet silently fixing bugs in your code. Blessed be the bits.
Terretta
... before being absorbed into his own AI and is now ascended in the Ethernet plane silently generating endless frames of international scandi-noir thriller serials for Netflix.
glonq
I'd love to borrow FB's brain for one week so that I could get an entire year's worth of work done.
jorvi
“Even in death, I serve the Omnissiah.”
deafpolygon
Chuck Norris once opened Windows and created the Macintosh.
ChuckNorris89
That's not true. When I opened Windows the telemetry disappeared.
nix23
Xerox...it was xerox for both.
dotancohen
Chuck Norris once opened Windows and no bugs came in.
galleywest200
He also wrote QEMU, which is probably far more impressive.
digianarchist
MuffinFlavored
This is missing QuickJS :) https://bellard.org/
xwolfi
Sorry, windows 98 running in JS is way way WAY more impressive. It's just insane.
mobilio
Also LZEXE back in time...
mhh__
The most impressive thing is the LTE modem
jraph
We probably should have a Fabrice Bellard accomplishments auto-comment on HN that starts by "Did you know…?" on those submissions.
MuffinFlavored
If we studied Fabrice from the perspective of "how can all be more like him/as productive as him/as innovative/as smart", do you think the findings would basically come back "he's just genetically gifted"?
He's like the Lebron James of the tech world from what I can deduct.
brigade
He’s good at creating useful MVPs that are architected well enough to attract other people to finish the remaining 99% of the work as he moves on to the next project.
astrange
ffmpeg is largely written by Michael Niedermayer if it's anyone.
The_Colonel
He's been gone from ffmpeg for many years.
Macha
He still appears to be the fourth highest contributor for 2022 according to the github mirror?
awill
This is ridiculous. Not only is Bellard no longer involved, but to say 'written' implies he did this by himself.
You wouldn't say Linux Torvalds wrote Linux. It's literally written by thousands of people.
I work with people like this, and I hate it when people say "I wrote X feature." No, no you didn't. It was written by a whole team.
lxe
I don't see anything wrong with this. It's like saying "Apple is built by Steve Jobs" even though obviously a lot more people are involved, and even if Jobs is no longer even around.
awill
No. That's not a good comparison. No one would say that about a company. A product, yes, but not a company. Steve Jobs founded, and led Apple.
hashim
Relevant - a question I asked on this topic not long ago: https://stackoverflow.com/questions/74828879/how-much-of-mod....
gyan
Changelog:
* Radiance HDR image support
* ddagrab (Desktop Duplication) video capture filter
* ffmpeg -shortest_buf_duration option
* ffmpeg now requires threading to be built
* ffmpeg now runs every muxer in a separate thread
* Add new mode to cropdetect filter to detect crop-area based on motion vectors and edges
* VAAPI decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
* WBMP (Wireless Application Protocol Bitmap) image format
* a3dscope filter
* bonk decoder and demuxer
* Micronas SC-4 audio decoder
* LAF demuxer
* APAC decoder and demuxer
* Media 100i decoders
* DTS to PTS reorder bsf
* ViewQuest VQC decoder
* backgroundkey filter
* nvenc AV1 encoding support
* MediaCodec decoder via NDKMediaCodec
* MediaCodec encoder
* oneVPL support for QSV
* QSV AV1 encoder
* QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
* showcwt multimedia filter
* corr video filter
* adrc audio filter
* afdelaysrc audio filter
* WADY DPCM decoder and demuxer
* CBD2 DPCM decoder
* ssim360 video filter
* ffmpeg CLI new options: -stats_enc_pre[_fmt], -stats_enc_post[_fmt],
* -stats_mux_pre[_fmt]
* hstack_vaapi, vstack_vaapi and xstack_vaapi filters
* XMD ADPCM decoder and demuxer
* media100 to mjpegb bsf
* ffmpeg CLI new option: -fix_sub_duration_heartbeat
* WavArc decoder and demuxer
* CrystalHD decoders deprecated
* SDNS demuxer
* RKA decoder and demuxer
* filtergraph syntax in ffmpeg CLI now supports passing file contents as option values, by prefixing option name with '/'
* hstack_qsv, vstack_qsv and xstack_qsv filters
exhilaration
This mostly looks like they're adding support for new codecs, there's only 4 new options in there, which seems rather paltry for a dot zero version release. Should we consider ffmpeg feature complete?
gyan
Dot zero releases basically signify major bumps of the core libs, and are not really synced with user-facing feature changes.
cm2187
I am still using 4.2. Does it make any difference if all I care about is h264 and hevc?
bsenftner
At a prior job we pinned to 4.2.3 because after that version an internal change massively impacted high frame rates. With a 1080p mp4 file on disk, and that disc being an ssd, version 4.2.3 delivers 700-800 frames per second, while all the versions after that 4.2.3 release - no other app or library changes - that same file may reach 300 fps. I spent some time trying to find the cause of that 50% cut in efficiency, but never found enough time.
Curious to try this version.
LtdJorge
More modern x264/x265 I guess
FranOntanaya
Nostalgia moment with the CrystalHD decoder deprecation. That came along the Intel Atom netbooks, one of the few ways we had to bump a bit their austere performance.
arjvik
I absolutely love ffmpeg, but for the life of me I cannot understand how its pipeline system works.
Each time I need to use it, I attempt to construct the command myself, but end up giving up and consulting StackOverflow. Amazingly, someone has usually done the exact thing I need to do and posted their command line to StackOverflow, so I'm never out of luck!
How do I actually start understanding how ffmpeg works? I want to be an ffmpeg power user.
xbmcuser
I know a lot of people look down on ChatGpt. But I have been using it for creating scripts to use with ffmpeg and I was able to get most of the things that I needed with very little massaging required. You can then ask ChatGpt about what it did and why so it will explain it as well so you can get some basic understanding on how things work. People don't realise what kind of tool Chat Gpt is and how to properly utilise it but it can be very useful for stuff like this.
amadvance
It really works! With few interactions, ChatGPT was able to create a not so obvious -filter_complex pipeline, like:
ffmpeg -ss 00:01:00 -t 00:02:00 -i input1.mp4 -ss 00:03:00 -t 00:02:30 -i input2.mp4 -filter_complex "[0:v][0:a][1:v][1:a]concat=n=2:v=1:a=1[v][a];[v]eq=brightness=0.3[outv]" -map "[outv]" output.mp4m3kw9
I asked gpt3 and here is what it said:
This command uses FFmpeg to create a single output video file from two input video files. It starts by specifying a start and duration for each of the two files (input1.mp4 and input2.mp4). It then applies a filter complex to the two files, which combines the two videos and audio into one stream, and adds a brightness filter with a value of 0.3. Finally, it maps the output video stream to the output file (output.mp4).
pjc50
And what does all that filter_complex do?
nicky0
Yes, chat GPT excels at comprehending and explaining things that have a consistent structure, restructuring, and and synthesising variations. If you keep it in its lane, it’s an excellent tool.
It’s really really bad at counting though. For example, try asking it to produce a line of 40 asterisks.
gamegoblin
It’s bad at counting because counting relies on a stateful O(N) algorithm you run in your brain.
GPT is trained to reproduce human text, which tends to simply have the output of this O(N) counting process, but not the process itself. So GPT “thinks” it should be able to just spit out the number just like human text implies we do. It doesn’t know we are relying on an offline O(N) algorithm.
If you have it emit a numbered list of 40 elements, it will succeed, because producing a numbered list embeds the O(N) process and state into the text, which is the only thing it can see and reason about.
Sunspark
It gave me 40 short Gaulish warriors..
beepbooptheory
Just be sure to thank all those stackoverflow repliers for this!
quijoteuniv
Agree! I used ChatGPT to explain some (uncommented)ffmpeg scripts I wrote few years ago. Scripts where created by going trough many websites and adapting to my needs. Explanation from chatGPT was spot on.
ly3xqhl8g9
Use ChatGPT, prompt: ffmpeg pipeline capture webcam to mp4
ChatGPT: >>>
ffmpeg -f v4l2 -framerate 30 -video_size 640x480 -i /dev/video0 -c:v libx264 -preset veryfast -crf 28 -pix_fmt yuv420p output.mp4
Here's a brief explanation of what each part of the command does:
- `f v4l2`: specifies that the input device is a V4L2 (Video for Linux 2) device, which is typically what webcams use.
- `framerate 30`: sets the framerate to 30 frames per second.
- `video_size 640x480`: sets the video resolution to 640x480 pixels. You can adjust this to your desired resolution.
- `i /dev/video0``: specifies the input device to use, which is typically /dev/video0 for the first connected webcam. If you have multiple webcams, you may need to adjust this to the appropriate device.
- `c:v libx264``: sets the video codec to use, which is H.264 in this case.
- `preset veryfast``: sets the encoding speed preset to "veryfast", which is a good balance between speed and quality.
- `crf 28`: sets the video quality using the Constant Rate Factor (CRF) method, with a value of 28. Lower values mean higher quality and larger file sizes, while higher values mean lower quality and smaller file sizes. You can adjust this to your desired quality/file size.
- `pix_fmt yuv420p``: sets the pixel format to YUV 4:2:0, which is a commonly used format for H.264 video.
- `output.mp4`: specifies the output file name and format.
<<<Never used the `crf` or `pix_fmt` flags personally, but seems correct (the issue with ChatGPT).
thrdbndndn
crf: "Constant Rate Factor" for x264 among other encoders. Think it like a output quality (higher = lower quality). I'd say 28 is way too high especially when you're using preset=veryfast (too bad quality). Since it's only 640-480, most of computers can handle fast or even medium in real-time. If not sure, I'd say just use default (by not giving anything).
pix_fmt: pixel format. 99% videos use this format, and it has best compatibility. FFMPEG will try to keep the original format when do transcoding, so in most of time you don't need to specify. However, in this very case, since the input is from a webcam, the chance it uses some weird format is high. And you don't want to keep that in your final result. So it's good to specify it out here.
My 2c: ChatGPT is great, but I recommend to read the comments it gives about each parameter, try to understand their purposes, and adjust accordingly if needed.
Also having a rough idea about how FFMPEG pipeline works (mainly the order of input, output and their associated switches in arguments) helps a lot.
Video process is a very complex thing and lots of time it relies on experience. Just be prepared sometimes your "typically works" command would break.
chx
> Never used the `crf` or `pix_fmt` flags personally, but seems correct (
Let me translate: the AI might or might not be bullshitting, I have no way to know but I decide to believe in it. Why ?
ly3xqhl8g9
Why do you trust the descriptions of `ffmpeg -help`? What if some evil daemon [1] went into the binary and changed completely the behaviour of the flags? Do you read the source code and verify the checksums for every program you run? In the real world, good or bad, very few people care beyond a first level of trust: does it work for my current issue? great, no?, try something else.
Also, you disingenuously left out the parentheses from the quote: using a tool requires undertaking its downsides, if the downsides can be mitigated accordingly then the tool is useful. Millions of users put to good use imperfect tools daily.
ttflee
You need `-pix_fmt yuv420p` for an iPhone to recognize a ffmpeg-encoded video file.
jeroenhd
Do you happen to know why? Is this a hardware decode limitation perhaps, or a bug in the standard decoder?
djkoolaide
and if you're encoding an x265 file, it needs
-tag:v hvc1
for iphone/quicktime to recognize it.
onion2k
You also need that same parameter for transparent videos.
izacus
Yeah, crf sets the quality and 28 will make your video look like a blurry over-compressed mess. Congrats :)
asicsp
Check out https://ffmpeg.guide/ (https://news.ycombinator.com/item?id=33383023)
See also https://img.ly/blog/ultimate-guide-to-ffmpeg/ (https://news.ycombinator.com/item?id=33771445)
pjc50
I was looking for ffmpeg.guide the other day and couldn't remember what it was called, thanks!
geerlingguy
I wouldn't call myself a power user by any means, but for many parts, the documentation is quite thorough, and if you're wrestling with some specific filter long enough, you might even begin to understand some of the magic incantation required to get it to work.
It's helpful to have some background in media container formats, compression algorithms, sound formats, and all the jargon and acronyms associated with the above. Easy!
rollcat
I know this doesn't answer your question, but have a look at GStreamer pipelines. They take the basic idea of shell pipelines, but add typing, format negotiation, M:N connections, etc all while giving you optional control over low-level details (which demuxer?), but also high-level abstractions (just show me a window with a preview). Once prototyped on the CLI (gst-launch-1.0[1]), they're also very easy to start using within an application through the framework (gst_parse_launch[2]), where you can e.g. iteratively add things like a volume slider. You can also access most of FFmpeg's codecs through libav.
[1]: https://gstreamer.freedesktop.org/documentation/tutorials/ba...
[2]: https://gstreamer.freedesktop.org/documentation/gstreamer/gs...
oldstrangers
Have you tried https://handbrake.fr/?
Aperocky
The ordering of the arguments changes behavior, I don't think that's a good thing or desired outcome, at most it's a bug_feature.
It tend to happen when big project rolls out batteries itself, in this case command line parsing.
gyan
The ffmpeg cli supports multiple inputs and multiple outputs, so there needs to be a way to unambiguously map an option to its target. Position is looked at, to group options for the same target together.
dylan604
argument order is definitely a feature. with -ss being one that behaves differently depending on it's location in reference to -i. it's not an accidental thing and the desired outcome dictates where you place it. not understanding that just means you're not using it enough to grok it.
Aperocky
Oh I used it enough to understand.
But I'm not convinced, maybe it is the local optima though.
orbisvicis
Even given an option it can be difficult to find the corresponding documentation, if only because of the many different submodules and encoders and decoders and filters that have oh-so-slightly different options. That said, I've just switched from pydub to ffmpeg-python (due to memory issues of the former[1]) and judging from the Jupyter notebook[2] it seems a much more intuitive method of constructing ffmpeg pipelines.
[1] https://github.com/jiaaro/pydub/issues/135
[2] https://github.com/kkroening/ffmpeg-python/tree/master/examp...
scottlamb
From the changelog [1]:
> - DTS to PTS reorder bsf
Interesting, I wonder what this is / why you'd want it. In particular, when you have the DTS but not the PTS.
The recent gstreamer 1.22 release [2] had what I read as the opposite—calculate a plausible DTS from the order and PTS. They did a nice job of explaining why it's useful. AFAICT, this approach is the only viable way to get B frames to work properly from a received RTP stream.
> H.264/H.265 timestamp correction elements ... Muxers are often picky and need proper PTS/DTS timestamps set on the input buffers, but that can be a problem if the encoded input media stream comes from a source that doesn't provide proper signalling of DTS, such as is often the case for RTP, RTSP and WebRTC streams or Matroska container files. Theoretically parsers should be able to fix this up, but it would probably require fairly invasive changes in the parsers, so two new elements h264timestamper and h265timestamper bridge the gap in the meantime and can reconstruct missing PTS/DTS.
Looks like the ffmpeg thing is dts2pts_bsf.c. [3] I haven't really read the implementation, but I was hoping the comment at the top would illuminate things, but "Derive PTS by reordering DTS from supported streams" isn't enough for me.
[1] https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/refs/heads/rel...
[2] https://gstreamer.freedesktop.org/releases/1.22/
[3] https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/refs/heads/rel...
Nexxxeh
To save anyone else the Google search:
> Decoding Time Stamp (DTS) and Presentation Time Stamp (PTS)
Snswered on Stack Overflow by slhck who linked to this tutorial:
http://dranger.com/ffmpeg/tutorial05.html
( https://stackoverflow.com/questions/6044330/ffmpeg-c-what-ar... )
jbk
This bsf was done to fix the following ticket https://trac.ffmpeg.org/ticket/502
scottlamb
Thanks. Looks like the input in question was an .avi file. That's a container format I don't know anything about (and don't particularly care to). I suspect then it's not relevant for anything more modern.
brigade
A surprising number of mp4 files are missing the ctts atom that contains pts since it’s sort-of optional outside the Apple stack (like ffmpeg generated such files from raw h264, which is probably the actual motivation for finally fixing this.) This should allow you to not generate or fix such files.
It’s sort-of optional for most playback stacks because they leave frame reordering to individual decoders as a codec-specific implementation detail, but Apple’s stack actually cares about frame-accurate random access so it relies on the codec-independent container timestamps.
The inaccurate seeking you get without container pts is okay for playback but it falls apart with editing or stuff like av1an.
QuarterRoy
I love software that just does what it’s supposed to do and is so good at it that it’s an underpinning for other projects.
FFmpeg, curl, nmap, et., al.
Aperocky
It is such a swiss knife kind of tool, there's just almost nothing it can't do when it comes to video, audio and image modifications.
I wonder how many big companies uses it "in secret" that we don't know off, or distributes it with mentions buried deep underneath usage agreement.
hnlmorg
It’s used heavily by professional broadcasters and TV/movie studios. AWS run it, satellite TV services use it, some VFX studios use it, etc. it’s become pretty foundational industry tech.
bretticus
And even the defense industry.
shmerl
Google must be using it for Youtube. Wikimedia is using it as well.
jzombie
I believe it is embedded in Chromium as well.
komali2
They do, they say so here: https://support.google.com/youtube/answer/1722171#zippy=%2Cc...
> YouTube does not recommend the RGB color matrix on uploads. In this case, YouTube initially sets the color matrix to unspecified before the standardization. It will then infer the color matrix using the color primaries during standardization. Note that sRGB TRC will convert to BT.709 TRC. YouTube re-tags the color primaries/matrix/TRC to BT.709 when it is not supported by FFmpeg colorspace conversion filter.
Since FFmpeg is gpl2.1 I thought they had to make it easier to know they're using this, like under a "licenses" section, but I don't see anything under studio.youtube.com indicating this.
voltaireodactyl
I can tell you it’s heavily used in professional settings within the entertainment industry. There is no better tool, thankfully!
tech234a
YouTube appears to have used ffmpeg, at least in 2011: https://news.ycombinator.com/item?id=2831292
c0nsumer
Do any of you know if this has Dolby AC-4 support? Ticket 8349 [1] has been open for years to add this, but it's not there yet. This would be very nice so we can watch ATSC 3.0 OTA broadcasts via FFmpeg-based things like Plex.
(Currently if one uses something like a SiliconDust HDHomeRun, viewing an ATSC 3.0 stream requires using their app/player, which uses a SiliconDust cloud service to do the decoding. It'd be really nice to have a not-network-dependent way to view/hear OTA broadcasts.)
ggm
ffmpeg is my go-to re-encoder. I didn't know ffprobe existed until I read some "howto" and its also increadibly useful as a way to get fundamental video and audio stream data out as e.g. csv or json.
So I can run an ffprobe script to get x,y info out, decide if the video needs re-encoding, pass to an ffmpeg call which does fast or veryfast settings to reset the x/y scale (for instance)
It's also unquestionably 'self documenting' because all of the sheharazade 1001 options are listed in --help. The problem is knowing which one will make the horse speak.
yboris
I've been so frustrated that FFprobe functionality is not part of FFmpeg.
My app extracts screenshots from videos to create a beautiful gallery of videos. But even though I include FFmpeg already, I need a 50mb FFprobe executable to be bundled with my app just so that I can determine the width, height, duration, and fps of a video file!
What is it that FFprobe does that FFmpeg couldn't do with a few extra pieces of exposed API?
https://videohubapp.com/ - https://github.com/whyboris/Video-Hub-App
https://github.com/whyboris/Video-Hub-App/blob/772b25bbd4b41...
jasomill
ffmpeg and ffprobe are built against the exact same set of libraries, so they could certainly be combined into a single program if the ffmpeg maintainers chose to do so.
Two possible options to reduce the size of your application:
(1) Instead of using ffprobe, just call "ffmpeg -i <filename>" without specifying an output file, then parse stderr:
$ ffmpeg -i https://download.dolby.com/us/en/test-tones/dolby-atmos-trailer_amaze_1080.mp4 >/dev/null
ffmpeg version 4.4.2 Copyright (c) 2000-2021 the FFmpeg developers
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'https://download.dolby.com/us/en/test-tones/dolby-atmos-trailer_amaze_1080.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf58.76.100
Duration: 00:01:03.55, start: 0.000000, bitrate: 4537 kb/s
Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 128 kb/s (default)
Metadata:
handler_name : sound handler
vendor_id : [0][0][0][0]
Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 4404 kb/s, 24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
Metadata:
handler_name : video handler
vendor_id : [0][0][0][0]
At least one output file must be specified
This is admittedly messy compared to parsing structured ffprobe output, but it does contain all the information you mentioned (assuming duration in centiseconds is sufficiently precise for your application).(2) Link both ffmpeg and ffprobe dynamically, in which case they'll share all but a few hundred kilobytes of on-disk code.
For example, consider ffmpeg and ffprobe as installed from package repositories on a variety of systems:
ffmpeg 4.4.2 installed by MacPorts on macOS Monterey (x86_64):
$ du -hA /opt/local/bin/ff{mpeg,probe}
339K /opt/local/bin/ffmpeg
260K /opt/local/bin/ffprobe
$ diff -s <(otool -L /opt/local/bin/ffmpeg) <(otool -L /opt/local/bin/ffprobe)
1c1
< /opt/local/bin/ffmpeg:
---
> /opt/local/bin/ffprobe:
72d71
< /usr/lib/libobjc.A.dylib (compatibility version 1.0.0, current version 228.0.0)
ffmpeg 5.1.2 installed from RPM Fusion on Fedora 37 (x86_64): $ du -h /usr/bin/ff{mpeg,probe}
284K /usr/bin/ffmpeg
176K /usr/bin/ffprobe
$ diff -s <(ldd /usr/bin/ffmpeg | awk '{ print $1 }') <(ldd /usr/bin/ffprobe | awk '{ print $1 }')
Files /dev/fd/63 and /dev/fd/62 are identical
ffmpeg 4.3.5 installed from raspberrypi.org on Debian 11.6 (aarch64): $ du -h /usr/bin/ff{mpeg,probe}
276K /usr/bin/ffmpeg
176K /usr/bin/ffprobe
$ diff -s <(ldd /usr/bin/ffmpeg | awk '{ print $1 }') <(ldd /usr/bin/ffprobe | awk '{ print $1 }')
Files /dev/fd/63 and /dev/fd/62 are identical
MinGW-w64 ffmpeg 4.4.3 installed by MSYS2 on Windows 10 (x86_64): $ du -h /mingw64/bin/ff{mpeg,probe}.exe
320K /mingw64/bin/ffmpeg.exe
184K /mingw64/bin/ffprobe.exe
$ diff -s <(Dependencies -depth 1 -modules /mingw64/bin/ffmpeg.exe | tail -n +2) <(Dependencies -depth 1 -modules /mingw64/bin/ffprobe.exe | tail -n +2)
Files /dev/fd/63 and /dev/fd/62 are identical
("Dependencies" is Dependencies.exe from https://github.com/lucasg/Dependencies)nickjj
I love ffmpeg, a single command of `ffmpeg -ss 01:15:42 -to 01:16:00 -i example.mp4 -c copy output.mp4` will let you create a video clip at a certain point in time from a larger video without decoding/encoding it.
I wrote a blog post and made a demo video the other day going over using this feature at: https://nickjanetakis.com/blog/create-video-clips-with-ffmpe...
vigneshv_psg
i understand that this is the fastest way to extract frames, but it is limited by key frame availability in the original file (i.e.) you can only start the output from a keyframe.
in this case, it is important to be aware that the times you specify may not be extracted exactly. it will be off by a few frames based on keyframe availability. the only way to extract exact frames is to re-encode. :)
nickjj
This is true. Do you have a use case where being off by a few frames might be a deal breaker?
Personally I've created dozens of clips using this method and it always turns out ok. It gives you about ~1 second precision on where you want to make your cuts. After I create the clips I can play things back normally, complete with an ability to seek to specific points successfully.
angrais
>> Do you have a use case where being off by a few frames might be a deal breaker?
Yes, in action recognition tasks (machine learning), e.g., if you have a large video with temporal annotations (start/end times where an action occurs) you may want to extract clips to sense-check the annotations. Being exact is important.
vigneshv_psg
yeah it probably doesn't matter in practice. you may run into some audio sync issues.
but the ~1 second precision that you see is by accident where the source file happens to have a keyframe every 1 second. that may not be the case always. :)
scottlamb
Fundamentally (ignoring ffmpeg incantations), I think there are four options:
1. re-encode the whole thing
2. re-encode just the first GOP
3. start a bit early or late
4. include the full first GOP but use an edit list to instruct the player to skip to the timestamp of interest
vigneshv_psg
yeah these are pretty much the options. i believe the ffmpeg command that OP suggested would do option 3.
sorenjan
I don't know if there's any work being done on this, but I wish ffmpeg had better support for modern ML based filters, like super resolution, frame interpolation, segmentation, automatic subtitles, etc. There was an ml filter made years ago as part of a Google summer of code projekt, which includes super resolution, but it's difficult to use and you need to train or find pretrained model files. ML is where video and audio filter research is happening at the moment, hopefully ffmpeg can get a good pipeline going. And please use an inference library that can run on all computers.
shmerl
What's the story with VAAPI AV1 encoding? Does it exist anywhere or not yet?
stonogo
It exists only as a patch from Intel (https://github.com/intel/cartwheel-ffmpeg/blob/master/patche...) but I haven't seen any efforts to upstream it yet.
shmerl
Thanks! At least changes look recent so there must be some plan to upstream it.
FrostKiwi
> nvenc AV1 encoding support > QSV AV1 encoder
That is captial 'A' A W E S O M E. FFmpeg is always on the bleeding edge, love it.
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If you want to know a bit more about what is important in this major release, you can watch my presentation about FFmpeg 6.0 at Fosdem:
https://fosdem.org/2023/schedule/event/om_vlc/ (Video + Slides)
It’s a bit less dry than the changelog, notably for the evolutions of the APIs.
What’s also important is the changes about the release schedule that we’ve been pushing with the community. Major version every year at the beginning of the year, with ABI and API break, minor releases during the year and an LTS every other year…