Re: [FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

[Cley Faye]

Le ven. 22 juin 2018 à 15:10, Christoph Gerstbauer <
christophgerstba...@gmail.com> a écrit :

> >
> >
> > ​The "-s" option is incorrect in your case. From the documentation:
> > > As an output option, this inserts the scale video filter to the end of
> > the corresponding filtergraph. Please use the scale filter directly to
> > insert it at the beginning or some other place.​
> >
> > The scale video filter will take whatever ends up from your filter, and
> > scale it without respect to aspect ratio.
> > To do what you want, you probable have to use a complex filter, where you
> > create a secondary black source and paste the output of your current
> filter
> > into it using the "overlay" filter.
> >
>
> I understand
> So i need to generate (after the cropping and deinterlacing to 720x576 50p)
> a filter syntax (within the videofilter) which transforms the 720x576 4:3
> content to a pillarbox 1050x576 quare pixel video (with black bars at
> sides) -> and THIS one I can rescale to HD.
> Correct?
>
> if yes, has anyone more experience with that?
>

​Roughly. Look at the other answer; it basically does the pillarbox *after*
the initlal rescale, which is probably a better idea.​
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Re: [FFmpeg-user] Unable to initialize SDL, over X11 for only root

[Cley Faye]

>
>
> In this particular case, there is a program daemon which is run as root
> that calls ffplay.  This is why the call is being made by the root user.
> It seems something is different in the case of root over X11 vs. a direct
> terminal call from the root user's desktop, but I don't have the foggiest
> idea.
>
> Any thoughts would be greatly appreciated.
>
>
>
​First thought is "can root run anything else using X11 in the same
condition". Instead of ffplay, try running something like xclock/xmessage
to check (there are probably better way to check this though).​
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Re: [FFmpeg-user] ffmpeg sometimes does not properly close the file

[Cley Faye]

2018-03-20 16:16 GMT+01:00 rbfmp...@t-online.de :

> I do a screencast with ffmpeg. The call looks like
> ffmpeg -f x11grab -video_size 800x600 -framerate 25 -i :2 -c:v libx264
> -pix_fmt yuv420p -preset ultrafast test.mp4
>
> The screencast is stopped by a SIGTERM (15) signal to the
> ffmpeg process. In most cases, this works fine.
> Occasionally, the ffmpeg process does not properly write
> the trailer to the file. Then the file is unusable.
>
>


> I have the following questions:
> 1. Does anyone know these exit codes, and in
> particular, does this give us a hint, what is
> going wrong?
>

​You might want to/should/must post the complete output from ffmpeg here.
If there's any hint at something going wrong, it will be in this log. It
will also provide information about what you requested, what ffmpeg
understood, what it saw, etc.​




> 3. Is it possible to repair the file? I corrupted a
> file by cutting off some kilobytes from a healthy
> mp4 file and tried to repair it with ffmpeg  copy.
> However, I get an error about an unprocessable
> file.


​MP4 files are not good at this. Some tools pretend to be able to fix such
broken files, but this is not very reliable. If possible, use another, more
fault-tolerant container such as mkv or even avi.​
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Re: [FFmpeg-user] probleme lors de la compilation de ffmpeg

[Cley Faye]

Le 7 mars 2018 à 12:35, CHRISTOPHE CBRO  a écrit :

> bonjour, pour l'école de musique de jeune enfants, nous essayons de
> fabriquer un mur de led pour les concerts.
> lors de la compilation de FFMPEG, nous avons des erreurs bloquantes.
> nous avons un raspberry Pi 3 mis a jour sous Rasbian.
>
> voici le config.log
>
> Accepteriez-vous de nous aider ?
>
>
>
​As far as I know, this mailing list is only in english.

The above roughly translate to "we get error when building ffmpeg on a rpi3
w/ raspbian, here's the config.log".​

The direct cause of this is that you ask configure to enable librtmp with
--enable-librtmp but the configure script can't find the development files
for librtmp. If you really want to support this, on a debian system
installing "librtmp-dev" should be enough if you're not doing anything
specific like cross-compiling or using custom prefix.

Note that this kind of error is likely to happen with other libraries if
you don't have all dependencies installed. Again, for most of them it'll
boild down to installing the appropriate "-dev" package *FOR SIMPLE CASES*.
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Re: [FFmpeg-user] FFmpeg dumps audio raw data of more size than expected

[Cley Faye]

2018-01-29 14:16 GMT+01:00 m.kamalasubha m.kamalasubha <
m.kamalasu...@gmail.com>:

> Hello all,
> I need to generate audio raw buffer for some 30 seconds from a clip whose
> sample rate is 44100 Hz. I used the following FFmpeg command to dump the
> raw data.
>   ffmpeg -i  -ss 00:00:00  -t 00:00:30 -f s16le -acodec pcm_s16le
> -ac 1 -ar 44100 audi.raw
>
> Expected audio raw file size : 44100 * 2 * 30 = 2646000
> Actual audio raw file size : 2646176
>
> I am wondering from where that extra 176 bytes arrive for 30 secs samples
> with sample rate 44100 Hz and sample size 2 bytes respectively.
>
>
>
​You really should post the complete output from ffmpeg here, it's the
*only* way to know what might be happening when the unexpected happens :)

(for the record, I tried your command and the output was as expected).​
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Re: [FFmpeg-user] Random repetitions in output

[Cley Faye]

2017-11-05 21:14 GMT+01:00 Mikael Persson :

> I cannot believe that there is no way to tell ffmpeg to extract all frames
> without repetition or dropping, shouldnt that be the default, and the
> behaviour I see a bug?
>


​There's no way to tell without knowing what happen, and there's no way to
know what happen without the full command and console output. For all I
know it could be as simple as a framerate mismatch.​or as complex as who
knows what.
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Re: [FFmpeg-user] Concat movies with different timebase

[Cley Faye]

2017-11-01 8:34 GMT+01:00 T4ng10r :

>
> I'm using this script to convert movies (insert timestamp) ->
> https://gist.github.com/T4ng10r/4f57212465a32172e4a4bb7a654fcd0e


​You should post here a single call to ffmpeg with all arguments, and the
complete output it gives​ (for a problematic case). This would probably
help a lot in providing an answer.
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Re: [FFmpeg-user] Why is concat so slow on a powerful machine?

[Cley Faye]

2017-10-18 3:17 GMT+02:00 Kevin Duffey :

> Yup. I do not specify the output. I just use the same .mov file extension
> as the two source files. Is it possible even though I specify -c copy
> out.mov it is rendering and not Concat??
>
>
>
​You really should provide a minimal command line reproducing the issue and
the output of ffmpeg, otherwise it will be impossible to divine what's
happening.

Or, if you want a general answer, no, concat should be fast assuming that
I/O are fast and the output muxer is fast. Without more information it is
hard to say anything more.​
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Re: [FFmpeg-user] Dolby E decoder

[Cley Faye]

>
>
> > >> ffmpeg -i input.wav -c:a copy -f u8 out.u8
>


> > > ffmpeg -i - -c:a copy -f u24le - | \
>


> > > The Dolby-E is 24bit, not 8.
> >
> > Doesn't matter, your example will not work, try it.
>
> Why doesn't it matter?
>
>
​When you copy a stream as-is, copying each byte individually or grouping
them by 3 doesn't matter. The "copy"​ codec simply pass the bytes making up
the stream without interpretation. Since all streams are made of bytes (at
least when put in a file), -c copy -f u8 will always work, whatever the
actual stream might be.

​This is just about the copy codec; why the remainder fail I have no idea.​
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Re: [FFmpeg-user] mp4 files size problem

[Cley Faye]

2017-09-07 15:57 GMT+02:00 Daniel <5960...@gmail.com>:

> Hello Everyone,
>
> I have tried to convert some videos to mp4 format what has surprised me
> was the size of the videos, After the conversion some videos were 2,3,4
> times bigger then the original wmv files. Here is the command i used to
> convert :  /usr/local/bin/ffmpeg -i video500.wmv -c:a libfdk_aac -c:v
> nvenc_h264  video500.mp4 ; After this command the mp4 file is 630MB  and
> the wmv file is 150MB (you can find attached the ffprobe to bothe files).
> If you can suggest any way on how to keep the quality and make the mp4
> files smaller please let me know .
>
>
> ps: On some mp4 videos the bitrate has doubled have no idea why ?
>

​If you don't supply your own settings, ffmpeg (and/or the underlying
codec) will assume some default values, not necessarily in compliance with
the input. In your case, you probably want (at least) to set the bitrate.
Also, I think that hardware encoder tend to produce bigger file for same
quality (and worse quality for same size), but this might not be accurate
anymore.​
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Re: [FFmpeg-user] Can ffpmeg output to stdout?

[Cley Faye]

2017-09-03 17:16 GMT+02:00 Mike Brown :

> ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on May  7 2013 01:09:00 with gcc 4.7.3 (GCC)
>

​That's very, very old. But probably irrelevant to the problem at hand.​



> Output #0, wav, to 'pipe:':
>   Metadata:
> ISFT: Lavf55.4.101
> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
> 5.1(side), s16, 4608 kb/s
> Stream mapping:
>   Stream #0:1 -> #0:0 (ac3 -> pcm_s16le)
> Press [q] to stop, [?] for help
> size=  243306kB time=00:07:12.54 bitrate=4608.0kbits/s
> video:0kB audio:243306kB subtitle:0 global headers:0kB muxing overhead
> 0.41%
>
>
​ffmpeg IS outputting to stdout as far as I can tell.​



> FFDCAENC --- experimental 'Coherent Acoustics' compressor.
>
> Input:   -
> Output:  ABC-HD-DD51-AudioTest.dts
> Bitrate: 1509 kbps
>
> Could not open or parse "-".
> Error: Data chunk not found!
>
>
>
​I believe your issue does not lay in ffmpeg but in ffdcaenc that can not
open stdin, as it plainly say he can't parse "-". To check this, you could
test by running the ffmpeg command by itself (which should just fill your
terminal with clutter and beeps). You can also try piping from a file to
ffdcaenc to make sure it work.

But as far as ffmpeg is concerned, it is outputting to stdout from that log.
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Re: [FFmpeg-user] flac format

[Cley Faye]

>
> If you convert the WAV to an other format you will have an approximation
> of the original data with the limitations of the codec of choice. FLAC
> claims to be lossless and offering about 50% storage savings.
>
> Storage is inexpensive why change the format and lose the original?


For the record, FLAC is lossless: going wav -> flac -> wav recreate the
exact same file, byte by byte.

I can see a few cases where smaller size for the same data is better: it
use less storage (obviously), transfer faster over internet/over other
networks, less memory bandwidth with hardware decoder (granted FLAC
hardware decoder are not that common). I see no advantage, when possible,
to not compress stuff, especially when the process is very fast in both
directions. Processing power is inexpensive too :)
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Re: [FFmpeg-user] flac format

[Cley Faye]

>
> Thanks.  Also the viceversa is as much simpler?


​For audio conversion with no change to layouts etc. it is usually that
simple (but don't trust this blindly; I'm sure one can find subtle issues
if one look hard enough)​



>   And, please, what about wma
> ​ ​
> format?  Is it also lossless?

I saw that the conversion wav > wma and wma >
> ​ ​
> wav is simple as well...
>

​WMA, aside from being proprietary, have multiple variations. There is a
lossless WMA format, but there's no guarantee it will work with anything.
For example, my local installation of ffmpeg can decode WMA Lossless, but
not encode it:

 D.A..S wmalossless  Windows Media Audio Lossless

 D.A.L. wmapro   Windows Media Audio 9 Professional

 DEA.L. wmav1Windows Media Audio 1

 DEA.L. wmav2Windows Media Audio 2

 D.A.L. wmavoice Windows Media Audio Voice

​​
​If your local installation happens to be able to use it, you'll have to
specify on the command line that you want to use 'wmalossless'​ in addition
to the file name. Otherwise it might use another (lossy) version.
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Re: [FFmpeg-user] flac format

[Cley Faye]

2017-09-01 10:55 GMT+02:00 Rodolfo Medina :

> Hi to all ffmpeg users.
>
> In order to backup store my audio CDs, I collect them into wav files using
> icedax, because I read that wav format is lossless.  Now, I've just read
> that
> also flac format would be, with the advantage of occupying much less memory
> space than wav.  As far as you listers know, is that true?  If so, what
> should
> the magic formula be to losslessly convert wav to flac and viceversa...?


​Yes, FLAC is lossless. Using ffmpeg to convert from wav to flac is pretty
straightforward:

$ ffmpeg -i input.wav output.flac

can't be much simpler than that :)
Of course you can specify a lot of flags for encoding, but by default it
will try to match the input, which is probably what you want.
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Re: [FFmpeg-user] Wav to Flac conversion: Invalid data found when processing input

[Cley Faye]

2017-08-20 10:56 GMT+02:00 Raman :

> input.wav 
>

​No idea if this is related, but this file isn't a .wav (or PCM) file.​
ffprobe and ffplay detect it as an mkv containing opus audio, and since it
is properly decoded, I assume it is.

It's a good idea to provide the full console output of ffmpeg as-is when
you get an error.
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Re: [FFmpeg-user] Quick (I hope) CLI question

[Cley Faye]

2017-07-24 22:24 GMT+02:00 Eric Downing :

> Hi!
>
> I have a folder of mixed format video files. I would like to:
>
> • copy video format
> • apply "ffmpeg-normalize" to the audio track(s), whilst preserving
> encoding
> • batch process so I can run the same command on the root folder
>
> For example:
> video1.mp4 (aac 5.1)
> video2.mkv (2 ch stereo)
>
>
> The issue is that I have a bunch of video files in varying video and audio
> formats that need loudness normalization. They number into the hundreds so
> running a single command to accommodate each files particular encoding is
> cumbersome. I want to preserve formatting for audio and video and apply
> normalization to the audio - in batch. Surely there must be a way to script
> this.
>
> I have found and understand the CLI language to do these things one by one,
> however I'm having a hard time concatenating these commands into a one
> liner. It seems that each time the CLI wants me to define an output format
> for the audio.
>
> Thank you for any suggestions.
>

​I'm not sure what you mean exactly, but if I understand correctly, here
are some pieces that might help you:

- If you don't specify an output codec (for either video or audio), it will
use some default settings
- If you specify "-c:a copy" (for audio) or "-c:v copy" (for video), you
can't use filtering as it will copy the stream as-is
- You can get file informations using ffprobe, in a script-friendly format
(like json)

By combining that, you can make a script that take a file, pass it to
ffprobe, get the audio codec and settings, and replicate an ffmpeg command
line that will copy video "-c:v copy" and use the appropriate audio codec.
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Re: [FFmpeg-user] man ffmpeg absent

[Cley Faye]

2017-07-18 16:26 GMT+02:00 Reindl Harald :

>
>
> Am 18.07.2017 um 16:17 schrieb Phil Rhodes:
>
>> Most of the people you are talking to are at least amateur software
>> engineers
>>
>
> what's wrong with you?


​I don't think it was meant as an insult (like amateurish), but more like
"enthusiast". At least it sounded like that to me as a non-native english
speaker.
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Re: [FFmpeg-user] man ffmpeg absent

[Cley Faye]

2017-07-17 23:32 GMT+02:00 ILJA SHEBALIN :

> What advantage does the bundle have over bare bone binary executables
> (UNIX execs in this case)? Do they enhance my experience supplementing
> basic features with extra ones?


​For starter, using a decent package manager provides a mean to install man
pages, which was the initial topic of this thread :)

man pages are just files that are put in a place where "man" (the software)
looks for them. They are not related to the actual executable file.​ If you
download an archive with only the executables, you get only the
executables. If you want the man pages, either you grab them and put them
in their place manually, or you let a package manager do it for you (with
the added benefit of having a way to update things at a later time).
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Re: [FFmpeg-user] scaling up without losing definition

[Cley Faye]

Le 18 juil. 2017 05:06, "JD"  a écrit :

Can ffmpeg do that?
Like from say 854x356 and transcode it to 1920x1080
without losing high definition (i.e. no blurring as a
result of transcoding to a larger display).


Ffmpeg can do scaling using various algorithm, including the "simple"
nearest neighbor that will upscale pixels nicely.

There are two caveats on that however: first 1920x1080 is not a multiple of
854x356, meaning that you'll either have black bars, cropped parts, or
imperfect scaling. Second, the actual video codec might mangle pixels
because of compression, and cause blurriness and other artifacts.
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Re: [FFmpeg-user] Issue when encoding a MP4 video

[Cley Faye]

2017-07-12 16:21 GMT+02:00 Léonce ANDRIANJAFY :

> Here is the complete output of the command:
>
> ffmpeg version 0.8.20-6:0.8.20-0+deb7u1, Copyright (c) 2000-2014 the Libav
> developers
>   built on Jan 19 2017 11:13:36 with gcc 4.7.2
> The ffmpeg program is only provided for script compatibility and will be
> removed
> in a future release. It has been deprecated in the Libav project to allow
> for
> incompatible command line syntax improvements in its replacement called
> avconv
>

​You are not using ffmpeg, you're using another product that masquerade
itself as an ffmpeg replacement using less than honest claims of
deprecation (to say the least).
​
You can get ffmpeg from https://www.ffmpeg.org/download.html , the exact
mean to do so depend on your system.
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Re: [FFmpeg-user] about merging more than 1 videos

[Cley Faye]

2017-07-06 10:27 GMT+02:00 akash kulkarni :

> Hello there,
> i am using the ffmpeg -copy command and merging multiple files from text
> file
> without re-encoding but its giving me jerk at the joint point of two files.
> How could i vanish that jerk.


​This depends on your source material (codec, settings of each files), how
they were made/cut, and on the exact command line you're using with ffmpeg.
Without any info, it's hard to tell.

For reference, I can stitch together multiple video without hiccups and
without doing much more than put their name in a file, use concat, and set
codec to copy.
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Re: [FFmpeg-user] Create hvc1 mp4 with ffmpeg only

[Cley Faye]

2017-07-03 5:43 GMT+02:00 桃源老師 :

> I also know that ffmpeg can produce hvc1 mov with the following command:
> ffmpeg -i hev1.mp4 -c: copy -tag:v hvc1 hvc1.mov
>
> Above information (hvc1 with mov) was found by Googling:
> https://ffmpeg.org/pipermail/ffmpeg-devel/2017-June/212538.html
>
> But if it is possible, I'd like to create hvc1 mp4 using ffmpeg only.  Is
> it possible?
>

​No idea, but have you *tried* doing this:
ffmpeg -i hev1.mp4 -c:copy -tag:v hvc1 hvc1.mp4​

?
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Re: [FFmpeg-user] Using FFMPEG to encode multiple outputs in 'parallel' (doesn't work?)

[Cley Faye]

2017-06-28 22:56 GMT+02:00 shalin :

> Piping only shows that I am trying to run all three ffmpeg transcode
> instances in parallel using single command line. If it makes any clearer,
> you can read them as three separate commands issued at the same time. With
> that, I hope we can focus on the concern raised in the original question.
>

​Then you don't want pipes, you just want three parallel processes, which
you can do using &.

More to the point, as far as I know, ffmpeg will not do parallel treatment
on either input or output, as you deduced. It certainly is doable, but
don't seem trivial to implement in the ffmpeg CLI tool itself since that
tool must handle various cases, and introducing (more) special cases would
add complexity for not much benefits.

You should look in using the various ffmpeg libraries directly; with that
you could probably write it to decode once, encode multiple time. But
unless you have shockingly similar settings and unlimited processing power,
you'll most likely stall some of your encodings while other catch up with
them. I don't know how well ffmpeg libraries handle multithreading (I
suppose fairly well since most of it's operations are based on context
structures) but you'll have to do a fair amount of synchronization by hand.
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Re: [FFmpeg-user] config log during ios for setup ijkmediaplay library

[Cley Faye]

2017-06-19 14:47 GMT+02:00 Alpesh Prajapati :

> PFA config.log file.
>
> Please help me out.
>

​Please ask a question first.

Also, have a look at this: https://trac.ffmpeg.org/wiki/MailingListEtiquette
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Re: [FFmpeg-user] id3 tags not really removed?

[Cley Faye]

2017-05-31 12:37 GMT+02:00 Rodolfo Medina :

> Further tests show that changes in tags are actually detected by the
> reader but
> with a delay: in a first moment they are not, then I `play' again with tags
> using ffmpeg and then the reader finally reads them...
>

​A shot in the dark here since I didn't have the patience to look at how
-write_id3v1 work; but if I remember correctly, id3v1 tags are actually
appended at the end of the mp3 stream in such a way that some older player
would even play them, causing a small burst of audio at the end of such
file.

If ffmpeg really doesn't handle them when reading the stream, it would
explain why 1- they get copied during stream copy, 2- why older tag would
still show up first, and maybe also 3- why the new tag would show up after
initial playing of the file, if the reader actually move from the end of
the file to detect them after reading.

What I'd suggest first is to try transcoding mp3 to mp3 instead of using
stream copy, just to make sure this actually destroy the existing id3v1
tags. If it does, then ffmpeg most likely just ignore these tag when
reading, making any attempt to "just" modify them kinda futile when copying.

(also, on your output file, you can open the file with some form of text
editor and jump at the end of the file, to check if multiple TAG section
are present. They'll be mostly in plain text.
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Re: [FFmpeg-user] ffmpeg: gif to video conversion does not keep gif time delay between frames.

[Cley Faye]

2017-05-07 18:16 GMT+02:00 Email Me :

> ffmpeg -f gif -i test.gif -f mp4 out2.mp4


​It's hard to tell without seeing the full output from ffmpeg, but there's
a distinct possibility you need to specify a framerate for the output using
-r.
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Re: [FFmpeg-user] embedded text subtitles

[Cley Faye]

2017-05-01 18:36 GMT+02:00 Gonzalo Garramuño :

> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'AnimSchool Animation Student
> Showcase 2015-OMYtIlwH3DY.mov':
>Metadata:
>  major_brand : qt
>  minor_version   : 512
>  compatible_brands: qt
>  encoder : Lavf56.40.101
>Duration: 00:02:52.78, start: 0.00, bitrate: 3315 kb/s
>  Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 3252 kb/s, 24 fps, 24 tbr, 12288 tbn, 4
> ult)
>  Metadata:
>handler_name: DataHandler
>encoder : Lavc56.60.100 libx264
>  Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
> stereo, fltp, 63 kb/s (default)
>  Metadata:
>handler_name: DataHandler
>

​There are no subtitles in that file, as pointed out by the part I quoted
(and confirmed by downloading the file).
If you see subtitles in vlc on windows (which I don't), they come from
somewhere else. Either it's a different file entirely, or there's a file
with the same name and the .srt extension in the same directory, which VLC​
​
​ will happily pick automatically.​
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Re: [FFmpeg-user] Memory usage growing when using ffmpeg. Is this expected?

[Cley Faye]

2017-04-28 16:52 GMT+02:00 Guilherme de Oliveira Costa <
guilherme.olive...@autotrac.com.br>:

> I am using ffmpeg record a video from a webcam, but the memory usage is
> very high, and keeps increasing with video length.
>
> For example, I use an embedded device with 64 MB of RAM, and when I
> recorded a 30 minute video, I was left with 2MB of RAM in the last minutes
> of the recording.
>

​I'm wondering, how do you check how much "free" memory you have while
ffmpeg is running? Since it didn't OOM or die, it might just have been
cache from the OS, which will gracefully yield if another program request
it.
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Re: [FFmpeg-user] Extract xvid stream in mov container

[Cley Faye]

2017-04-28 8:18 GMT+02:00 jacques perconte :

> [mpeg4 @ 0x7ffc82817800] header damaged
> [mpeg4 @ 0x7ffc82817800] time_increment_bits 4 is invalid in relation
> to the current bitstream, this is likely caused by a missing VOL
> header
> [mpeg4 @ 0x7ffc82817800] time_increment_bits set to 5 bits, based on
> bitstream analysis
> [mpeg4 @ 0x7ffc82817800] looks like this file was encoded with
> (divx4/(old)xvid/opendivx) -> forcing low_delay flag
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7ffc82800400] decoding for stream 0 failed
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7ffc82800400] Could not find codec
> parameters for stream 0 (Video: mpeg4 (xvid / 0x64697678),
> yuv420p(bt709), 10032 kb/s): unspecified size
>
> Consider increasing the value for the 'analyzeduration' and 'probesize'
> options
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'HYPERS.MOV':
>
>   Metadata:
> major_brand : qt
> minor_version   : 537199360
> compatible_brands: qt
> creation_time   : 2015-03-09T00:01:04.00Z
> com.apple.quicktime.player.movie.visual.brightness: 0.00
> com.apple.quicktime.player.movie.visual.color: 1.00
> com.apple.quicktime.player.movie.visual.tint: 0.00
> com.apple.quicktime.player.movie.visual.contrast: 1.00
> com.apple.quicktime.player.version: 7.6.6 (7.6.6)
> com.apple.quicktime.version: 7.7.3 (2826.24) 0x7738000 (Mac OS X,
> 10.9.5, 13F34)
>
>   Duration: 08:53:49.52, start: 0.00, bitrate: 10033 kb/s
> Stream #0:0(eng): Video: mpeg4 (xvid / 0x64697678),
> yuv420p(bt709), 10032 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc (default)
>

​From that, I'd say the input file is damaged and ffmpeg can't get the
video parameters, especially the actual layout and resolution of the video
stream.

Does the file play fine with ffplay or other players? Usually ffmpeg does a
good job of recovering some damages, but it vary from codec to codec.​
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Re: [FFmpeg-user] Extract xvid stream in mov container

[Cley Faye]

2017-04-25 9:52 GMT+02:00 jacques perconte :

>
> ffmpeg -i input.mov -c:v copy -an output.avi
> ffmpeg -loglevel debug -analyzeduration 6M  -i nput.mov -c copy -s
> 1920x1080 output.avi
>
> it always print
> "[avi @ 0x7fe9e303ce00] Dimensions not set
> Could not write header for output file #0 (incorrect codec parameters
> ?): Invalid argument"
>
> any idea ?


​Provide the full console output of these commands, it might give some
insight on what's failing.
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Re: [FFmpeg-user] Warning "Starting new cluster due to timestamp" on muxing two input streams and resulting movie with muted audio at some locations

[Cley Faye]

2017-04-24 11:22 GMT+02:00 Georgi D. Sotirov :

> What is the general reason for warning "Starting new cluster due to
> timestamp" to appear anyway?


​I'm not sure of the exact cause of this warning by ffmpeg, but if the
actual message is to be trusted (usually it is), it could be an
inconsistency in timestamps. Clusters are kind of an MKV "unit" that
contain the actual audio/video data (the frames). If there is a hole in the
timestamps, or if they are out of order, the muxer might create a new
cluster to accomodate this.​
​
​Since it only happen on the audio, you might want to check that the audio
stream itself is well formed, and also check the output file with some
other player (try ffplay)​
​
​. Other than that, a greater ffmpeg "guru" will have to step in ;)​
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Re: [FFmpeg-user] Warning "Starting new cluster due to timestamp" on muxing two input streams and resulting movie with muted audio at some locations

[Cley Faye]

2017-04-24 9:58 GMT+02:00 Georgi D. Sotirov <gdsoti...@dir.bg>:

> Anyone? Meanwhile I've done the muxing with mkvmerge from MKVToolNix <
> https://mkvtoolnix.download/> package without any problem and muxed other
> movies with FFmpeg successfully.
>

​Could you give us the full command line and output from ffmpeg?​


-- 
Cley Faye
http://cleyfaye.net
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Re: [FFmpeg-user] Problem with ffmpeg-3.2.4

[Cley Faye]

2017-04-13 11:18 GMT+02:00 Pierre Delmas <
pierre.del...@centrale-marseille.fr>:

>
> How to replace avcodec_decode_video2 in the version ffmpeg-3.2.4 ?


​Two things.

First, this is not the right list for this kind of questions;
https://lists.ffmpeg.org/mailman/listinfo/libav-user/ is probably a better
starting point.

Second, it might be a shot in the dark, but since the documentation (
https://www.ffmpeg.org/doxygen/3.2/group__lavc__decoding.html#ga3ac51525b7ad8bca4ced9f3446e96532)
is telling you to use avcodec_send_packet() and avcodec_receive_frame(),
maybe you could use avcodec_send_packet() and avcodec_receive_frame()? :)
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Re: [FFmpeg-user] concatenate .h264 through a file

[Cley Faye]

2017-04-11 12:07 GMT+02:00 bbo2 :

> Guys I am a noob apologies if this has been asked..
>
> I have over 900 files to concatenate and am using following command:
>
> ffmpeg -f concat -i files.txt -vf scale=1080:-1 -vcodec mpeg4 -b:v 1000k
> output.mp4
>
> I Created a txt list(files.txt) with names of the videos (video-1.h264 - to
> video-943.h264)
>
> the response I get is "[concat @ 00e967e0] Line 1: unknown keyword
> 'video-1.h264'
> files.txt: Invalid data found when processing input
>
> I am executing ffmpeg in the video directory... so don't need full path to
> videos, right?
>
> what am I doing wrong?
>

T
​he usual ​reply would be for you to post the actual content of files.txt
and the complete ffmpeg log.

However, I'll give you both these links in advance, showing you the full
help of the concat demuxer as well as a proper input file in this case:
https://ffmpeg.org/ffmpeg-formats.html#concat
https://trac.ffmpeg.org/wiki/Concatenate

:)
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Re: [FFmpeg-user] Converting and encoding in same step without pipes

[Cley Faye]

2017-04-06 5:31 GMT+02:00 Aarthi Priya Thirumalai <
aar...@multicorewareinc.com>:

> I want to convert my input from Prores 4K10 bit 4:2:2 to 4K10bit 4:2:0 and
> directly encode through libx265/libx264 in ffmpeg in a single step instead
> of doing the conversion and storing in a intermediary file and again
> feeding that into ffmpeg.
>
> I would like to know if there is a way to do this  - conversion and
> encoding in a single step, without using system pipes in ffpmeg? pls let me
> know your thoughts and suggestions.
>

​Not sure if a filter would fit your requirements, but if you're using
pipes you're not creating intermediary files.
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Re: [FFmpeg-user] Cross compiling ffmpeg lib with openh264

[Cley Faye]

2017-04-06 13:21 GMT+02:00 Shivam Raj :

> I am trying to compile with Cyclone-V SOC with Yocto toolchain for ARM
> cortexA9. I am receiving the following error :
> ERROR: openh264 not found using pkg-config
>
> ./configure  --enable-cross-compile  --cross-prefix=arm-poky-linux-
> gnueabi-
> --arch=arm --target-os=linux --prefix=/home/it-0118/arm_crosscompiled_bin
> --enable-static  --enable-libopenh264
> --extra-cflags="-I/home/it-0118/arm_crosscompiled_bin/include/
> -L/home/it-0118/arm_crosscompiled_bin/lib/ -lopenh264"
> --extra-ldflags="-L/home/it-0118/arm_crosscompiled_bin/lib"
>
> I have attached the config.log file with the mail.
>
>
Since you're cross-compiling, you need two things: pkg-config that will
look in the correct place to find library built for your target​
architecture, and libraries actually built for that architecture, which are
different from the one installed on your build system.

I see two things in your log:

> WARNING: arm-poky-linux-gnueabi-pkg-config not found, library detection
may fail.

Meaning it will use the "regular" pkg-config which might not know where are
the library for your build target, and ultimately

​> ERROR: openh264 not found using pkg-config​

Meaning pkg-config did not find openh264 at all.

Make sure you have the development files for openh264 for your build target
appropriately installed, try to cross-compile a simple example using them,
then try to use them building ffmpeg.

 The information in this message may be confidential and is being sent

> exclusively to the receiver. If you are not the recipient of this
>

​This email was sent to a publicly archived mailing list...​
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Re: [FFmpeg-user] Shrinking video files

[Cley Faye]

2017-04-04 15:25 GMT+02:00 Cecil Westerhof :

>
> Yes, that is what I did. People can of-course download the videos, but
> it would really be nice when they just could view them immediately in
> the web-browser. H.264 is significantly bigger as H.265, but I think
> the user experience is more important as saving space on Google Drive.


​It might be worth trying encoding your files in with VP9 for video then.
It might give you better results (space-wise) than h264 and works
relatively well in browsers (
https://developer.mozilla.org/en-US/docs/Web/HTML/Supported_media_formats#Browser_compatibility
)
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Re: [FFmpeg-user] Starting to copy takes long

[Cley Faye]

2017-03-29 1:54 GMT+02:00 Cecil Westerhof :

> I want to copy a little bit of a video. I do this with:
> ffmpeg -i input.mov -ss 16:28 -t 5 output.mov
>
> But it takes about 2:20 before the copying starts. Is this normal?
>

​Try moving the -ss part before your input. -ss before an input use forward
(if possible fast forward) on your source, while before an output it means
processing everything before, then start using the output after the given
time.
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Re: [FFmpeg-user] ffserver support

[Cley Faye]

2017-03-26 18:21 GMT+02:00 Mickael MONSIEUR :

> It's a dammage to see ffserver abandoned. I hope your message will wake up
> some passionate :)
>

​To my knowledge, last time the issue was raised, it wasn't abandonned:

https://ffmpeg.org/pipermail/ffmpeg-user/2017-February/035158.html
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Re: [FFmpeg-user] Convert iPhone MOV to Android MP4

[Cley Faye]

2017-03-10 7:14 GMT+01:00 Richard Duran :

> I will have to try that. I was about to revert back to my original
> question, wondering why going from AAC input to AAC output didn't retain
> the input bitrate.
>

​(all of the following is what I think, knowing how I use ffmpeg)​

B
​ecause​ it doesn't make much sense from ffmpeg's point of view to even
know the input codec settings when creating output.

First, some codec settings aren't necessarily present in the input anyway.
I don't know for AAC, but it's more obvious for video codec where a lot of
encoding tweaking isn't visible in the output metadata. Reusing what is
known (bitrate and samplerate) without all other settings would probably
produce something different, and might or might not be "the best way" to
retain the best quality from the source material.
Add to that that an "output" (from ffmpeg's point of view) might not even
come from anything: the output of a filterchain for example.

Second, and I guess this is more obvious, transcoding with lossy codec WILL
lose source data. Assuming all settings are equal, transcoding from a
100kbit/s AAC stream to a 100kbit/s AAC stream will lose some data from the
source. Even transcoding at a higher bitrate, you will lose something
(although it might not be noticeable the first few times you do it). That's
the nature of lossy compression (same thing apply to video codec, jpeg
images, etc.)

Third, ffmpeg can't tell your intent. Since, because of my first point, it
can't reuse the input settings, it does what seems to be the best next
choice: use sensible settings to keep good quality. This part is totally
subjective, and that's why we have so many settings. As far as I know, for
most codec, ffmpeg's default tends to be alright (I regularly transcode
video to x264 with no settings for quick previews, they come out ok). But
it can't tell if you want to keep the source data as much as possible, or
if you want to keep good quality (which will incur loss of data, but not
necessarily in a noticeable way), or if you want a quick transcode to test
something else.

​And finally, fourth, something I just thought while writting this: if you
want to transcode from AAC to AAC, with the same input settings, to keep
the same quality, it's not going to happen. Even if ffmpeg was modified to
retain the input codec settings automatically, and had all these
informations at the end, as long as you ask for a transcode it will do it,
meaning recompressing the input. You *really* want to use the copy method
here.

Writting a small script that take input files, probe them using ffprobe,
then create ffmpeg command with either "-c copy" or appropriate aac
settings seems to be the cleanest way you could achieve what you're asking.
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Re: [FFmpeg-user] Convert iPhone MOV to Android MP4

[Cley Faye]

2017-03-08 23:06 GMT+01:00 Richard Duran :

>
> Example #2, input with pcm_s16le, input bitrate of 705 kb/s. Output bitrate
> defaults to 69 kb/s. I would have expected the output bitrate to match that
> of the input. Does AAC support that high of a bitrate?


What are you trying ​to do and to what end? Matching the bitrates between
pcm (uncompressed) and AAC (lossy compression) doesn't make much sense.
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Re: [FFmpeg-user] Convert iPhone MOV to Android MP4

[Cley Faye]

2017-03-08 17:56 GMT+01:00 Richard Duran :

> From what I can tell of the following output, the input audio bitrate is 85
> kb/s, and the output is 69 kb/s.
>
> I would like to know if it's possible to retain the source audio bitrate. I
> am doing this via a batch file that loops through iPhone videos that have
> varying bitrates.
>
>
>
> E:\> ffmpeg.exe -i 100APPLE\IMG_0075.MOV -vcodec copy -acodec aac
> C:..\iPhone\DCIM\IMG_0075.MP4
>

​Since your input already have aac audio, and you're already copying the
video stream as-is, why not just "-codec copy"​ and be done with it?

If you specify "-acodec aac" it will do a transcode, taking time, and use
default values, which is obviously not what you want.
Also, if you do want to transcode from aac to aac, there's no much point in
trying to keep the same bitrate; it'll be lossy anyway.
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Re: [FFmpeg-user] -movflags faststart revisited

[Cley Faye]

2017-02-26 23:32 GMT+01:00 JD :

> Are there any flags that will let me playback an unfinished transcode?
> Reason I am asking is that if my various options in the transcode command
> did not yield what I like, then I would like to abort the transcode.
>

​mp4 files will never be playable without "finishing" them. But you can
interrupt the process at anytime with 'q'; that will properly close the
file so you can play it.​
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Re: [FFmpeg-user] -c:v or -vcodec copy or -codec:v

[Cley Faye]

2017-02-22 8:39 GMT+01:00 Tim Hiles :

> What is the deal with all the different ways to copy?  Do they all do the
> same thing? Cause if not, it is NOT clear what they are and if soseems
> like a lot of redundancy.
>

I
​'m not​ sure from what they come from, but they achieve the same thing.
From the documentation:

​-vcodec codec (output)
> Set the video codec. This is an alias for -codec:v.​


​-c is just a shortcut for -codec, and :v is used to apply the argument to
video streams.​
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Re: [FFmpeg-user] ffmpeg concat issue

[Cley Faye]

2017-02-17 23:49 GMT+01:00 Mark Hatch :

>
> Any thoughts?


​Hi,​

​You should​ probably send to the list the command *with* the complete
console output when it fails, in addition to what you expect to happen.
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Re: [FFmpeg-user] Extract all audio regardless of stream/channel count

[Cley Faye]

2017-02-10 12:40 GMT+01:00 Carles Vila :

> If you could give me a starting point, I could probably figure out.


​This comes with no guarantee, no documentation and very minimal testing,
but since you asked about a starting point...

I put up a small python script that take a file with many streams, detect
audio streams, output mono streams as single output file and stereo streams
as two files, all in one pass.
The script doesn't test for any cornercase, so there.

Even if it's not exactly fit for your task, it shows how you can easily
retrieve the output of ffprobe and produce a minimal ffmpeg command line.
http://pastebin.com/C53fqZG7
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Re: [FFmpeg-user] Suggestions for FFmpeg: support a simpler command line for concatenating audio files with a specified overlap

[Cley Faye]

2017-01-29 18:30 GMT+01:00 a :

> Hello, after hunting around ffmepg.org, I could not find where to post
> suggestions for FFmpeg developers, so if this mailing list is the
> appropriate place here you go otherwise please let me know:
>
>
>   1.  I have a need to concatenate programmatically several short (< 1
> second) mp3 files but with a specific overlap for example 100ms, the
> current solution is a command line of this form
>  *
> 1.  ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3
> 2. -filter_complex
> 3.   "[1]adelay=delay1[b];
> 4.[2]adelay=delay2[c];
> 5.[3]adelay=delay3[d];
> 6.[0][b][c][d]amix=4"
> 7.  out.mp3
>
>
>  *   However this forces me to derive each file durations and do the
> math.  I was hoping if you would consider supporting something that would
> not require retrieving the durations and just specify the relative delays
> during concatenation.  Here are a couple of examples
>
>  *   Delay all input file by the same amount
> ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3 -filter_complex "delay=100;
> amix=4"
>
>  *   Delay each input by the specified amount or zero if not specified.
> ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3 -filter_complex "[1]delay=100;
> [2]delay=20; amix=4"
>
>  *   Of course this may not be generalized enough to offer the most
> flexibility so I will let you decide on the final form of the options, but
> you get the point.
>
>  *   NOTE: I am using C-Sharp to build the command line for other
> FFmpeg tasks, however the simpler command line above would allow easier
> prototyping from a batch file without requiring hardcoding exact value for
> delay1, delay2,, delayN
>
>
I think you got your answer right there. ffmpeg is very flexible, but is
usually used with an "overlay" program to leverage more common use cases.
Even without using the bindings in various languages it's very powerful.​
Combined with the easily parseable output of ffprobe you can easily do
anything you want already. Also, changing the syntax of filter strings this
way would probably break way more things than it'd be helpful.
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Re: [FFmpeg-user] Why are absolute paths for concat unsafe?

[Cley Faye]

2017-01-20 14:45 GMT+01:00 Shriramana Sharma :

> But still I am not sure why exactly absolute paths are considered
> unsafe. I would like an explanation for this. Thank you!
>

​If I remember correctly, some people used ffmpeg in tools that allowed
user input to be part of the concat file list, allowing path traversal and
other nasty stuff.
It also disallow space in names, some characters, etc.
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Re: [FFmpeg-user] Troube with piping into ffmpeg and fast seeking

[Cley Faye]

2017-01-07 23:44 GMT+01:00 expenses :

> For example:When I run the above
> command
>

​I believe you forgot a critical part in your message :)

You should post the full command, and full output produced.
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Re: [FFmpeg-user] Why is length and bitrate zero?

[Cley Faye]

2016-12-18 21:57 GMT+01:00 Michael Heuberger <
michael.heuber...@binarykitchen.com>:

> ffmpeg -f image2 -thread_queue_size 2048 -framerate 18.9 -i
> /home/michael-heuberger/binarykitchen/code/videomail.io/var/
> local/tmp/clients/videomail.io/11e6-c1e3-fc2ad020-ae4a-
> 290f0f20d08f/frames/%d.webp
> -y -an -vcodec libvpx -filter:v
> drawtext=fontfile=/home/michael-heuberger/binarykitchen/code/
> videomail.io/src/assets/fonts/Varela-Regular.ttf:text=
> www.videomail.io:fontsize=10:fontcolor=0xFF@0.25:x=text_
> h:y=main_h-(text_h*2):shadowcolor=0x11@0.15:shadowx=1:
> shadowy=1,scale=trunc(iw/2)*2:trunc(ih/2)*2
> -crf 8 -pix_fmt yuv420p -deadline good -cpu-used 1 -loglevel warning
> /home/michael-heuberger/binarykitchen/code/videomail.io/var/
> local/uploads/videomail.io/videomail/11e6/c1/e3/11e6-
> c1e3-fc2ad020-ae4a-290f0f20d08f/videomail_good.webm
>

​Unless I missed something obvious here, the full output of ffmpeg
(including headers) will be required to be able to tell more.
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Re: [FFmpeg-user] FFMPEG 3.2.2 - "could not find codec parameters"

[Cley Faye]

2016-12-18 12:59 GMT+01:00 Caio Cesar Moreira :

> I'm trying to dump video frames from a TS file but for some reason ffmpeg
> is failing.
>
> Command used: ffmpeg -i playlist.m3u8 -vcodec -r 1 "output_%04d.png"
> -loglevel 99
>
>

> Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
> ('copy' to copy stream)) with argument '-r'.
>
> Reading option '1' ... matched as output url.
>

C
​an't tell about the rest of your message, but you must put a codec name
after -vcodec. As it is, your command mean "use codec​ '-r' for the video
of output named '1'", which is probably not what you want.
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Re: [FFmpeg-user] how to convert one mp4 file into multiple files with different setting

[Cley Faye]

2016-12-17 8:41 GMT+01:00 qw :

> I want to use ffmpeg to do the following complicated task:
>
>
> I have one mp4 file with one audio and one video stream. I want to convert
> the mp4 file into three encoding settings, and then save each output with
> one encoding setting into 3 files.
>
>
> How to do it by using single ffmpeg command?
>

​Every time you provide an output filename, all output settings are reset.
You could do something like this:

$ ffmpeg -i input.mp4 -c copy out1.mp4 -c:v h264 -c:a libvorbis out2.mp4 -c
copy -map 0:v out3.mp4

Each output will have the preceding options applied to them.
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Re: [FFmpeg-user] how to concatenate several flv files into one flv file

[Cley Faye]

2016-11-17 14:21 GMT+01:00 qw :

> >All methods and each of their restrictions are listed here:
> >https://trac.ffmpeg.org/wiki/Concatenate
>
>
> I use the following command, which doesn't work since vlc can play the 1st
> file but not 2nd file in the resulting flv file.
>
>
> ffmpeg -i "concat:file1.flv|file2.flv" -c copy -f flv file_mixed.flv
>
>
> does ffmpeg support the function of merging two flv files correctly?
>

The Concatenate page explicitely tells about this command. It works at the
file level, and only certain files can be concatenated this way. Use any of
the other method.
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Re: [FFmpeg-user] ffplay on current build does not play mkv file but on old build does.

[Cley Faye]

2016-11-10 20:12 GMT+01:00 Tim Hiles :

> I attached the file in a previous email but ffmpeg list rejected the email
> because it was 750 KB, so I'm sharing a google drive link to the file,
> otherwise I'm not really sure what else to do.
> https://drive.google.com/open?id=0B8g9J5RbDqElbU5RUDRoeVBmYXEzW
> HZ6SFdUWGhKR1MwYzJB
>

​For what it's worth, playing this file with both those zeranoe builds
works fine (second one is from today) :
ffplay version N-82178-g60178e7 Copyright (c) 2003-2016 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth
--enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
--enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264
--enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libsnappy --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg
--enable-lzma --enable-decklink --enable-zlib
  libavutil  55. 35.100 / 55. 35.100
  libavcodec 57. 65.100 / 57. 65.100
  libavformat57. 57.100 / 57. 57.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 66.100 /  6. 66.100
  libswscale  4.  3.100 /  4.  3.100
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100

ffplay version N-82314-g863ebe6 Copyright (c) 2003-2016 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth
--enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
--enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264
--enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libsnappy --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg
--enable-lzma --enable-decklink --enable-zlib
  libavutil  55. 36.100 / 55. 36.100
  libavcodec 57. 66.101 / 57. 66.101
  libavformat57. 57.100 / 57. 57.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 66.100 /  6. 66.100
  libswscale  4.  3.100 /  4.  3.100
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
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Re: [FFmpeg-user] Using ffmpeg to blur moving objects

[Cley Faye]

2016-10-31 15:23 GMT+01:00 Boris T :

> Yes, I already have the moving objects' coordinates (x, y values, the box's
> width and height, as well as the duration of the blur). I only need to use
> ffmpeg to blur that object in the video according to the provided
> coordinates. I have a working solution, but it becomes very slow the more
> blurred overlays I have.
>

​I am not sure I'm getting the whole picture of what you're doing, but if
you have a lot of blurring to do on each frames, maybe a script that output
each video frames into another tool doing the blur, which in turns output
into a second ffmpeg process for transcoding could avoid you a cascade of
overlays.
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Re: [FFmpeg-user] How to output the MPEG-4 streaming to file?

[Cley Faye]

> When I use command "ffmpeg -i udp://@{IpAddr}:{Port} -c:v copy -c:a copy
-f
> mpegts stream_dump.ts", i got the following results.
>   - H.264...OK.
>   - MPEG-2...OK.
>   - MPEG-4...NG. "Invalid data found when processing input" will occur.

What is required to better understand the issue is the full command and the
full output of the ffmpeg process, from the version header to the end. It
will most likely contain the required informations.
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Re: [FFmpeg-user] Configure does not find any dependencies anymore

[Cley Faye]

2016-10-10 19:57 GMT+02:00 Nomis101  :

> OK. Please find attached the config.log. The output of "pkg-config
> --list-all" is:
>
> $ pkg-config --list-all
> -bash: pkg-config: command not found
>

​Also in the config.log:
> WARNING: pkg-config not found, library detection may fail.​


​Others might be more knowledgeable about the specifics of mac dev.
environments, but it seems that pkg-config is not available.
There seem to be support for pkg-config in homebrew.
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Re: [FFmpeg-user] Configure does not find any dependencies anymore

[Cley Faye]

2016-10-09 22:37 GMT+02:00 Nomis101  :

> //ERROR: gnutls not found using pkg-config//
> //
> //If you think configure made a mistake, make sure you are using the
> latest//
> //version from Git.  If the latest version fails, report the problem to
> the//
> //ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on
> irc.freenode.net.//
> //Include the log file "config.log" produced by configure as this will
> help//
> //solve the problem./
>

​I believe you forgot to include the "config.log" file; it might contain
useful informations.

It would probably be interesting to also give the output of "pkg-config
--list-all", since configure seems to use that.
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Re: [FFmpeg-user] encoding video from png diffs

[Cley Faye]

2016-09-01 19:31 GMT+02:00 Joshua Grauman :

> I could of course generate all the pngs, but it would take up a lot of
> space and I'm wondering if anyone knows of any creative way to do this?
> Thanks!


​You could just do that. There is no need to write them anywhere though;
you can pipe png sequences into ffmpeg directly (I create animations this
way).

I'm not familiar with vcdiff, but as long as you have a tool that produce a
png on a standard output, you could loop over your diffs, piping the output
info ffmpeg. Maybe you'll need a temporary variable storing the resulting
png for the next step, but that will be held in RAM so no worries. Be
careful if you do that though; binary data and shell script variables are
not too friendly. I'd do that in python or another language.

Once you have your script that output the sequence of png on it's standard
output, you can pipe them to ffmpeg using '-' as the input name. Something
like this:
$ your_script | ffmpeg -f png_pipe -i - out.mp4
Don't forget to specify framerate etc.

(of course, if ffmpeg does support such format by itself, ignore this
message; I'm not familiar with *all* the stuff ffmpeg supports either).
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Re: [FFmpeg-user] What is Segmentation fault: 11 ?

[Cley Faye]

2016-08-22 23:56 GMT+02:00 Nomis101  :

> There you are: http://polysom.verilite.de/tmp/Test.mp4.zip
>

​For what it's worth, the issue is not present with the static 64-bit build
from zeranoe (not master but close). There are some messages that might be
related to this issue though:
[mp4 @ 00f17fe0] track 1: codec frame size is not set

Full log:
C>ffmpeg -i Test.mp4 -c copy -map 0 out.mp4
ffmpeg version N-80924-gb450b82 Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth
--enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enable-iconv --enable-libass --enable-libbluray --enable-libbs2b
--enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm
--enable-libilbc --enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil  55. 28.100 / 55. 28.100
  libavcodec 57. 48.103 / 57. 48.103
  libavformat57. 41.100 / 57. 41.100
  libavdevice57.  0.102 / 57.  0.102
  libavfilter 6. 47.100 /  6. 47.100
  libswscale  4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Test.mp4':
  Metadata:
major_brand : mp42
minor_version   : 512
compatible_brands: isomiso2mp41
creation_time   : 2016-08-22 21:50:00
encoder : HandBrake 0.10.5 2016021200
  Duration: 00:00:53.66, start: 0.00, bitrate: 1490 kb/s
Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D),
yuv420p, 504x272 [SAR 251:252 DAR 251:136], 206 kb/s, 25.15 fps, 100 tbr,
90k tbn, 42187 tbc (default)
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: VideoHandler
Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo,
fltp, 640 kb/s (default)
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: Stereo
Side data:
  audio service type: main
Stream #0:2(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo,
fltp, 640 kb/s
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: Stereo
Side data:
  audio service type: main
[mp4 @ 00f17fe0] Using AVStream.codec to pass codec parameters to
muxers is deprecated, use AVStream.codecpar instead.
Last message repeated 2 times
[mp4 @ 00f17fe0] track 1: codec frame size is not set
[mp4 @ 00f17fe0] track 2: codec frame size is not set
Output #0, mp4, to 'out.mp4':
  Metadata:
major_brand : mp42
minor_version   : 512
compatible_brands: isomiso2mp41
encoder : Lavf57.41.100
Stream #0:0(und): Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 504x272
[SAR 251:252 DAR 251:136], q=2-31, 206 kb/s, 25.15 fps, 100 tbr, 90k tbn,
90k tbc (default)
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: VideoHandler
Stream #0:1(und): Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz,
stereo, 640 kb/s (default)
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: Stereo
Side data:
  audio service type: main
Stream #0:2(und): Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz,
stereo, 640 kb/s
Metadata:
  creation_time   : 2016-08-22 21:50:00
  handler_name: Stereo
Side data:
  audio service type: main
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
  Stream #0:2 -> #0:2 (copy)
Press [q] to stop, [?] for help
frame= 1349 fps=0.0 q=-1.0 Lsize=9766kB time=00:00:53.66
bitrate=1490.8kbits/s speed=1.03e+003x
video:1355kB audio:8385kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.265055%
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Re: [FFmpeg-user] (no subject)

[Cley Faye]

2016-07-24 17:45 GMT+02:00 Walid Salman :

>
> [root@ns1 ~]
> ​​
> # ffmpeg -i /home/lamas44747/public_html/sarmaye.mp4 -ss
> 00:00:10 -to 00:00:20 -c copy /home/lamas44747/public_html/split.mp4
>
> the server show this error :
>
> FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
>   built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
>   configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64
> --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth
> --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
> -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC'
> --enable-avfilter --enable-avfilter-lavf --enable-libdc1394
> --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin
> --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger
> --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl
> --enable-nonfree --enable-postproc --enable-pthreads --enable-shared
> --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
>   libavutil 50.15. 1 / 50.15. 1
>   libavcodec52.72. 2 / 52.72. 2
>   libavformat   52.64. 2 / 52.64. 2
>   libavdevice   52. 2. 0 / 52. 2. 0
>   libavfilter1.19. 0 /  1.19. 0
>   libswscale 0.11. 0 /  0.11. 0
>   libpostproc   51. 2. 0 / 51. 2. 0
>
> Seems stream 0 codec frame rate differs from container frame rate: 48.00
> (705505/14698) -> 24.00 (705505/29396)
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
> '/home/lamas44747/public_html/sarmaye.mp4':
>   Metadata:
> major_brand : isom
> minor_version   : 512
> compatible_brands: isomiso2avc1mp41
> encoder : Lavf56.40.101
> comment : http://www.aparat.com/kshahrvandi
>   Duration: 00:00:58.88, start: 0.-46440, bitrate: 662 kb/s
> Stream #0.0(eng): Video: h264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], 562
> kb/s, 24 fps, 24 tbr, 705505.00 tbn, 48 tbc
> Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 96 kb/s
> Unrecognized option 'to'


​There are multiple ways to do this, but I assume in this case you want
your input to start at 00:00:10 and end at 00:00:20 from the input video ?
First, know that using copy, it is possible that the cut won't begin
exactly where you want it: to produce a readable output, a lot of codecs
requires the stream to start on an iframe. If you want exact cut, you might
need to reencode a part of your video.

Second, I believe that this time (I was wrong on this before), you want to
-ss and -to to be *input* options and not *output* options. With ffmpeg,
the argument order is important: any option related to an input or an
output must be provided before the input/output. If you have an option that
is related to an input, it must be before the corresponding "-i".
Also note that depending on which option you choose, the relative timestamp
of your input might change. The "-t" option allows you to specify a
duration instead of a timestamp if this is an issue.​

​If I'm not wrong, your command line would look something like this:​
​​# ffmpeg -ss 00:00:10 -to 00:00:20 -i
/home/lamas44747/public_html/sarmaye.mp4 -c copy
/home/lamas44747/public_html/split.mp4​

​Here, "-ss" and "-to" relate to the input file, "​-c copy" relate to the
output.
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Re: [FFmpeg-user] Need help resolving concat error

[Cley Faye]

2016-07-23 21:39 GMT+02:00 Mark Strecker :

> *ffmpeg -i files.txt -c:v copy -c:a copy OUT.MOV*
>
> *ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers*
>


> *Input #0, tty, from 'files.txt':*
>
> *  Duration: 00:00:00.04, bitrate: 29 kb/s*
>
> *Stream #0:0: Video: ansi, pal8, 640x400, 25 fps, 25 tbr, 25 tbn, 25
> tbc*
>


> *Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid argument*
>

​I think it would be interesting to see the content of files.txt? Maybe
it's missing the header, and since you didn't specify "-f concat" either,
it can't work with that.
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Re: [FFmpeg-user] Ffmpeg Concat Images failure

[Cley Faye]

2016-07-09 14:39 GMT+02:00 Vivek Jain :

> Below is the output. I see the below 2 files which errors out are PNG
> whereas others are JPEG. I guess that's an issue?
>

​I'm still learning about ffmpeg, but what I know is that concat expect all
the files to have roughly the same layout and codecs. As it is, it is
decoding all your files using what's found in the first file (a jpeg), so
all your files must be jpegs.
The same thing happen if you use concat with multiple files with clashing
audio settings (different rates, different channel layouts, etc): the
output is garbled (or just fail to process).

Also, although it doesn't matter here, you might want to avoid naming your
files ".png" if they really are jpegs...
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Re: [FFmpeg-user] Ffmpeg Concat Images failure

[Cley Faye]

2016-07-09 6:43 GMT+02:00 Vivek Jain :

> Please let me know if entire output or actual images are needed.. Much
> thanks...
>

​Entire output is always a good idea.
Also, I'm not sure, but I don't think 'duration' is used that way in a
concat file. Complete output might confirm this.
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Re: [FFmpeg-user] Duration Issues

[Cley Faye]

2016-06-30 17:15 GMT+02:00 Celso Junior :

> ffmpeg -i origin.mxf -ss 00:00:00.000 -t 00:00:01.000  -c copy -map 0 -y
> chunk.mxf
>
>

> I tried setting the fps by hand, using -to instead of -t but nothing
> changes...
>
> Does someone has some idea about it? What am I doing wrong?
>

​I'm not sure how it would produce the result you're seeing, but for
information input option (I assume -ss and -t refers to the input here)
must be placed before the corresponding -i.
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Re: [FFmpeg-user] Problem with using pipe:1 when creating webm audio

[Cley Faye]

2016-06-27 19:11 GMT+02:00 Mike Speciner :

> I'm attempting to create a webm audio on the fly within my website, with a
> command like
>
> ffmpeg -i one2twenty.wav -aq 6 -dash 1 -f webm pipe:1 > foo.d1p
>
> although the piped output is actually being sent out over the web; I just
> redirected it to file in the above so I could investigate it. If instead I
> do
>
> ffmpeg -i one2twenty.wav -aq 6 -dash 1 -f webm foo.d1f
>
> I get a similar file, but slightly shorter. The problem is that the piped
> version (foo.d1p) doesn't work properly in firefox--apparently there is
> something wrong with the cues and firefox (using the  element)
> claims the audio is not seekable and won't seek into it, although it will
> play it from the beginning. There is no such problem with the directly
> written version (foo.d1f).
>
> Do I have to do something special to get the piping to work, or do I have
> to fall back to writing the file and then reading it back in to serve it? I
> discovered this problem using ffmpeg 2.7.2, but I've now tried it with
> ffmpeg 3.1, and it still fails the same way.
>
> By the way, I tried to submit a bug report (assuming this problem is
> actually an ffmpeg bug), but when I attempted to register, the system
> claimed my registration was spam without giving me any clue as to why.
>

​I don't think it is really a bug in ffmpeg: the output using a pipe can't
obviously move back to update the header when the transcoding operation is
complete​ to create a seek table. Seeking in such a stream should (IMO)
rely on the player's abilities to build such seeking index.

If it's really a header problem, I don't think using a pipe to produce such
file can ever work; if I remember correctly, in a webm the seeking entries
are located before the actual streams and are updated after said streams
are written.
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Re: [FFmpeg-user] Why the fps of concatenate video file change with old video file?

[Cley Faye]

2016-06-23 11:12 GMT+02:00 cuizhe :

> >ffmpeg -r 25 -f concat -i 1.txt -c copy output.wmv
>
> 1.txt:
> file g:\1.wmv
> file g:\2.wmv
>
> It works. But the fps of output.wmv is 30.
> How can I maintain the fps of concatenate video file?
>

You should post the full log, it most likely contains the answer.
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-22 19:45 GMT+02:00 juan carlos Rebate :

> the problem is in ffmpeg not know what else wants me to say, I have
> indicated the error, as I indicated all screen output with the error, have
> not received any solution, if you want something specific let me know
>

​I, and other, pointed out that your command line was incorrect *and* that
your choice of codec/container was invalid for what you're trying to do.
Some provided *working* example.

There is no detailed explanation on the error code because it spells out
what's wrong:
> [mp4 @ 028d5840] muxer does not support non seekable output
> Could not write header for output file #0 (incorrect codec parameters ?):

It plain say "mp4 muxer does not support non seekable output". If you
insist on using a container that "does NOT support NON SEEKABLE output"
with a *non seekable* output such as a stream obviously you'll keep getting
an error.
What I learned is that if google don't give anything useful with an error
message, there is two possibilities: you've discovered a new problem, or
you're doing something wrong. Seeing that ffmpeg is used in that exact
scenario by a lot of people, well... I'm not saying there is no bug in
ffmpeg but "take input X, output to Y" is one of the feature that works
fine.

Granted, I don't have the whole picture, but to fix what you posted in this
thread, just changing the mp4 container to anything that support streaming
works fine, as said by others, as said by me, and as *showned* in a video.
I'm baffled why you keep saying "oh no there is no solution this program is
clearly not working" when you've got many already.

As far as I'm concerned, this thread is done: you've given a bogus command,
you've got answers as to what was wrong with it, and a working example. I'm
afraid there's nothing more that can be done by random people there.
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-22 18:36 GMT+02:00 juan carlos Rebate :

> ¿I can send photos or capture video ?, so'll see that does not work


Since you've been very uncooperative to *any* kind of help and always
filled every replies with attacks toward ffmpeg, and since this command
line have something like four arguments and you can't test it, I doubt
you'll believe a video showing it simply work. Anyway, it took mere seconds
to record, so here it is.
​
https://www.youtube.com/watch?v=JLz-0zMqlhM

Now, if you still believe that ffmpeg is completely unusable and can't do
the simplest task despite every attempt to help you (and show you!) on this
list (from people that have more patience than me), you'll have to wonder
if the problem really lies in ffmpeg.
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-22 17:51 GMT+02:00 juan carlos Rebate :

​​
> The same command line you provided, with matroska replacing mp4 works fine
> ​ ​
> for a standard input file:​
> >ffmpeg -i lp_e1m3_a.mkv -f matroska -c:v mpeg4 -listen 1
> ​ ​
> http://0.0.0.0:80


>


>
>
with this command vlc can not see the flow when you try to connect gives the
> error message I have indicated before, there is no way to make it work


​How do you try to connect with VLC? This command works fine with my
version of ffmpeg, which is already quite old:
> ffmpeg version N-78990-g689211d Copyright (c) 2000-2016 the FFmpeg
developers
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-22 0:05 GMT+02:00 juan carlos Rebate :

> I've tried all combinations for codecs mpeg4 libx264 also webm libvpx
> with the same negative results
>

The same command line you provided, with matroska replacing mp4 works fine
for a standard input file:​
>ffmpeg -i lp_e1m3_a.mkv -f matroska -c:v mpeg4 -listen 1 http://0.0.0.0:80

​Only differences here:
 - I'm using an input file, as setting an rtp server just to test this is
silly
 - I'm not providing an address to listen from
 - Using mp4 as a container gives the exact same error you have anyway

Using the above command, I was able to see the stream from VLC with no
issue.
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-21 23:35 GMT+02:00 juan carlos Rebate :

> Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid argument


​That's relatively clear to me. Have you tried *any* other combination of
codec/containers? As far as I can tell, ffmpeg won't stream mp4 containers.
Note that using network addresses does not mean "simply pipe the file on
the socket"...
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Re: [FFmpeg-user] Could not write header for output file #0 (incorrect codec parameters ?):

[Cley Faye]

2016-06-21 21:06 GMT+02:00 juan carlos Rebate :

> I keep looking but not quite find anything to help me solve the problem


​> ffmpeg -y -i rtp://239.0.0.107:8208

 -map 0 -f mp4 -c:v mpeg4 -listen 1
http://localhost:80  -multiple_requests 1 -seekable 1
-bufsize 200k -r 299
-re -b:v 982k​

​What was told in the previous message is that some of your arguments are
misplaced. In the quoted command line, everything after localhost:80 is
ignored because it is misplaced. Maybe fixing that first will help.​
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Re: [FFmpeg-user] What is the correct way to cut the video with ffmpeg?

[Cley Faye]

2016-06-21 17:44 GMT+02:00 桃源老師 :

> Still curious about cutting video on non key frame is correct or not...


​Depends on your definition of correct, but for my definition (which is, I
need an output file that play fine in most players) I'd say "no".
It varies between codecs of course, but "usually", if you don't reencode,
cutting anywhere else than a key frame will result in a garbled stream up
until the next key frame.
From this page: https://trac.ffmpeg.org/wiki/Seeking the behavior of ffmpeg
when seeking and *not* transcoding is to seek to an i-frame (key frame) and
adjust the stream timestamp accordingly, which might be why some players
are confused.

If you want accurate seeking outside of keyframes without some sort of
timestamp adjustment, you'll have to transcode the video/audio. You can
either transcode the whole segment, or just transcode up to the next
keyframe then copy from there but as far as I know you'll have to do it "by
hand".
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Re: [FFmpeg-user] Read mp4 video from pipe

[Cley Faye]

2016-06-17 12:21 GMT+02:00 kolychev :

> $ cat video.mp4 | ffmpeg -pix_fmt yuv420p -i pipe: -r 25 -f image2pipe
> -vcodec mjpeg -q:v 1 pipe: > result
>
> ffmpeg version 3.0.2 Copyright (c) 2000-2016 the FFmpeg developers
>   built with gcc 5.3.0 (GCC)
>   configuration: --prefix=/usr --disable-debug --disable-static
> --disable-stripping --enable-avisynth --enable-avresample
> --enable-fontconfig --enable-gnutls --enable-gpl --enable-ladspa
> --enable-libass --enable-libbluray --enable-libdcadec --enable-libfreetype
> --enable-libfribidi --enable-libgsm --enable-libiec61883
> --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb
> --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus
> --enable-libpulse --enable-libschroedinger --enable-libsoxr
> --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2
> --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxvid --enable-netcdf
> --enable-shared --enable-version3 --enable-x11grab
>   libavutil  55. 17.103 / 55. 17.103
>   libavcodec 57. 24.102 / 57. 24.102
>   libavformat57. 25.100 / 57. 25.100
>   libavdevice57.  0.101 / 57.  0.101
>   libavfilter 6. 31.100 /  6. 31.100
>   libavresample   3.  0.  0 /  3.  0.  0
>   libswscale  4.  0.100 /  4.  0.100
>   libswresample   2.  0.101 /  2.  0.101
>   libpostproc54.  0.100 / 54.  0.100
> Option pixel_format not found.
>

​I think the error message is explicit enough here. You're trying to apply
the option pix_fmt to an input file. Simply remove it (or move it after
your inputs and before your output if required) and it should work (at
least here it does).
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Re: [FFmpeg-user] How to re-encode 3d video to 2d video

[Cley Faye]

​2016-06-07 23:43 GMT+02:00 jd1008 :

> It really needs an installer that will put all those files into the
> appropriate
>
>> directories, same as how installing an rpm does.
>
>> As it is, I cannot (will not) use it because I will have to put it into a
> separate directory
> to which my PATH env will point, and I dont want to keep mucking with that.


​Feel free to make one ;)
RPM packages do exist, but I'm not sure how well this will work with FC20:
http://rpmfusion.org/
The remark about the need to update is that security fixes are most likely
not backported that far. That pose a security threat as many
vulnerabilities will apply to the system. Of course if it stay off the
internet and *always* get its input from safe source, whatever. F20 end of
life was (according to wikipedia) at the end of 2013.

Alternatively, it is not uncommon to have a ~/bin directory in your path,
and quite tidy too. Also, ffmpeg static builds don't need any extra file to
operate beside the binary; you only need the ffmpeg binary somewhere. You
can even drop it in your system binaries directory (something like
/usr/local/bin is made for that, at least in Debian-based systems).
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Re: [FFmpeg-user] cannot allocate memory

[Cley Faye]

2016-06-06 10:34 GMT+02:00 Madovsky :

> ok the last git (today) seems to solved the problem
> thanks
>

​Oooh, good! Thanks from me too! I can go back to weird mix-n-match of
filters using amix :)​
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Re: [FFmpeg-user] How to re-encode 3d video to 2d video

[Cley Faye]

Le 6 juin 2016 03:36, "jd1008"  a écrit :
>
> Is this the way to convert a 3d vid to 2d?
>
> Moritz had replied to another question of how to play 3d vid on computers
> not capable of playing 3d. So I used his -vf options as follows:
>
> ffmpeg -i the_3d_video.mp4 -r 29  -vf
"crop=w=iw/2:h=ih:x=0:y=0,scale=w=2*iw:h=ih,setdar=2"  -y the_2d_video.mp4
>
> The command above is still running, and will likely take hours on my slow
computer.
>

From your filter command I assume it's a 3D half side by side video. As far
as i know, any operation to turn one of the view into a full-resolution
video will need encoding and that process takes time. There are command
line options to speed up the process, but anything high quality will be
time consuming.

Your best bet here is to either find what settings you find acceptable to
balance quality and speed, or use a player that can fake it.
(With some container it might be possible to do that through metadata but
even so, finding a player that would understand it is another matter)
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Re: [FFmpeg-user] ffmpeg upgrading

[Cley Faye]

2016-06-03 17:05 GMT+02:00 Jase :

> I’ve been trying unsuccessfully trying to upgrade to the current version.
> I have followed any /all instructions I could find but to no avail and  I
> really need it updated so one of my sites will work properly . Please can
> some one help. Version and codecs installed are below.


​If you installed ffmpeg using your distributions' tools (things like
aptitude/dpkg or rpm, etc.) you can use that.
If you installed ffmpeg "by hand", simply replace the binaries; as far as I
know there is no "installation" process. If you're a simple user on a
system, you can also just drop ffmpeg's binaries in your path (~/bin is a
good place for that).

More details on what was unsuccessful on your end might help to provide a
more detailed answer.
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Re: [FFmpeg-user] FFMPEG | Error converting video to a thumbnail

[Cley Faye]

2016-05-28 18:34 GMT+02:00 C0nw0nk :

> ffmpeg.exe -y -analyzeduration 2147483647 -probesize 2147483647 -itsoffset
> -768 -i file.mp4 -vcodec mjpeg -vframes 1 -an -f rawvideo -s 178x100
> fileimage.jpg
>

​I'm still learning how to use ffmpeg efficiently, but I think you provide
way too much stuff in that command line if your intent is to only create a
single-frame jpeg from a video input.​
As far as I can tell, your input command works with "ffmpeg version
N-78990-g689211d".
Also, from your log, it seems that ffmpeg is struggling with your input
file:

> [h264 @ 0554b9e0] no frame!
> [mov,mp4,m4a,3gp,3g2,mj2 @ 04e3d300] stream 0, offset 0x1e79f03:
> partial file
> file.mp4: Invalid data found when processing input


My guess here is that the file is incomplete/corrupt.

I have some question about your command line though, to help me understand
ffmpeg a bit more:
- the analyzeduration and probesize seems superfluous (for the video stream
in an mp4 file anyway). Are they really needed?
- itsoffset might not be as good as -ss performance wise. At least in the
version I use, -ss immediately seek the asked time, while itsoffset crawls
the file until it reach the timestamp requested (ignore this question if
the input doesn't start at 0...)
- "-f rawvideo" and "-vcodec mjpeg" seems also superfluous, as ffmpeg will
happily output to jpeg if given a jpeg filename
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Re: [FFmpeg-user] stream mapping seems to work, but doesn't

[Cley Faye]

2016-05-24 18:17 GMT+02:00 sean darcy :

>
> Instead of trying to fix stream order, is there another easier way?


​It might not be the best option, but it should work: you could remux all
your VOBs into ​correctly-arranged mp4 (or any other container that know
better than VOB), then concatenate them. All this using -map 0:v and -map
0:a to avoid potential ordering issue for intermediate VOBs. This option
needs some space to work, but it should be quite fast and scriptable.
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Re: [FFmpeg-user] drawtext 4:3 vs 16:9

[Cley Faye]

2016-05-14 22:52 GMT+02:00 naser sonbaty :

> I need help with filter_complex and drawtext.
> drawtext on aspect 4:3 are correct but drawtext on aspect 16:9 are
> stretched.
>

​I've tried the drawtext filter with both 4/3 and 16/9, and in both case
the text was the same (pixel perfect even). No stretch issue.

It seems that in both your command, the video itself have the same
resolution (720x576); only the SAR and DAR change. My understanding on it
is that the DAR is only used for playback, but the SAR is the "real"
picture. In any case ffmpeg will write the same picture of text on both,
but the video player will stretch the input video to fit either 4/3 or
16/9, including the now hard text.

If you need to add text to a video with SAR/DAR that won't match, you have
to do it in a way so that the "stretched" output will look normal.
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Re: [FFmpeg-user] nvenc ffmpeg

[Cley Faye]

2016-05-16 2:06 GMT+02:00 gofrane :

> I use Nvidea GeForce GT 610 and I think I have  the  recent drivers
> installed ( else which is the recent version )
>

​It doesn't matter which driver version you got; the hardware does NOT seem
to support nvenc. If you want nvenc to work, you'll have to update the
hardware.

According to the linked articles, the oldest card that have the
capabilities to support nvenc would be a GT630, having a Kepler chip (but
don't quote me on that and do your research if you intend to buy hardware).
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Re: [FFmpeg-user] Prepend a single frame to a MP4 file

[Cley Faye]

2016-05-15 22:36 GMT+02:00 Virgil Stokes :

> I have created an image file (testImage.png) from which I would like to
> make a 5 second MP4 file. Then
> concatenate (prepend) this 5 second video to the test.mp4 file (as defined
> above). How can this be done in ffmpeg? Note, I am using:
>

​My approach with that would be to create a 5 second video from the image
(using something like "ffmpeg -loop 1 -framerate 29.97 -i  -c:v h264
-pix_fmt yuv420p out.mp4" then concatenate them using the concat input.
Assuming matching codecs and resolution, this would work well without
having to transcode the whole video. You might also need to add an empty
audio track to the 5s segment beforehand though.
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Re: [FFmpeg-user] FFMPEG run out of memory

[Cley Faye]

2016-05-09 1:06 GMT+02:00 ionut :
>
> Is this enough ? I need to say that I have a very large ( huge ) command
> that is being executed so a paste isn't good, and once I attaced to the
> output specification the '-report' and the report already had over 100k
> lines for 30s of output, while my total time is over 360.
>

​It would be better to post it anyway; text is not that heavy for e-mails :)
At least the full command and a big chunk of the output log would be good.
If you still think that the output log is too big, the most relevant parts
(in my opinion) is the beginning (everything from start to the beginning of
the transcode), the end (everything after the transcode progress until the
very end), and if something funny happens in the middle, everything
slightly before and slightly after, even lines containing supposedly known
informations: it is good to know that the program also know these.
In your case, it is most likely that ffmpeg output *something* before
crashing.​ And in any case, worst scenario is it's too much for anyone to
look at.


I also have some suggestions that might or might not be easy to implement
in your solution, that might (emphasis on "might"​) lower the memory usage,
in case it is the real culprit here. Again, note that my experience with
ffmpeg is mainly on the "I use it and it just work" side, so there:

 - If you only want no sound, maybe just not generating a soundtrack could
be a good idea

 - The background input have no size specified (default is 320x240 but I
doubt it suits you), and maybe setting the framerate inside the filter
instead of using -r can improve things (I also have a preference of putting
these things inside the same filter expression as the rest, which might or
might not help ffmpeg to better understand the filtergraph). By setting at
least the size I suppose you could remove some scale...

​- For each image input, since you feed them in a complex filter it might
be possible (quick testing says yes, but I don't know your whole situation)
to not have each images at 30 FPS. When I do some sort of collage, I end up
having each image having a rate of 1/duration, effectively making them a 1
frame source. The filtering ends up producing a video at the right
framerate so it's not an issue, but it might help.

​Note that none of these "advices" is likely to help for your warning
message, ​only with memory usage. Also, the third trick did yield a smaller
memory footprint (for a 1minute video and a single image source, going from
231MB to 226MB) alongside a *serious* boost in speed (40seconds to
16seconds). If anything, it could help your command to crash faster ;)
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Re: [FFmpeg-user] FFMPEG run out of memory

[Cley Faye]

2016-05-09 0:04 GMT+02:00 ionut :

> My output is .mp4 and I am using like 50-100 images to create it. Image
> resolutions are very big and I do a lot of complex filtering. I have some
> cases that my cmd run out of memory and the ffmpeg process is killed.
> One more thing,  I have the next warning:
> [png_pipe @ 0x1f9c480] Stream #0: not enough frames to estimate rate;
> consider increasing probesize
>
> But if I add some big analyze and probesize values like: ( saw in an
> example over the internet )
> -analyzeduration 6000M -probesize 6000M
> The process is killed even faster.
>

​I'm curious, ​it seem that you're piping the images into the ffmpeg
process, so I'm tempted to say you need to specify the framerate in the
command line. However, on a recent ffmpeg version it simply defaults to 25.
​Obviously setting such high probesize is probably never useful anyway...​


> Can somebody give me some tips about why is this happening ?
> I know that I haven't given you to many details, please hit me back for
> more details, thank you!
>
​​
​To begin with, the full command and output log would be useful. Preferably
a minimal working example if you have a really complex filter. The problem
might be on the way you provide the source images to ffmpeg (just a hunch).​
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Re: [FFmpeg-user] move from audio track to another

[Cley Faye]

2016-05-05 23:43 GMT+02:00 juan carlos Rebate :

> https://trac.ffmpeg.org/wiki/AudioChannelManipulation
>
>
> here it is perfectly explained this in version 2.8 and earlier applies to
> ffmpeg and ffmplay


​The link you provided explicitely say the same thing: -ac allows you to
set the *number* of output *channels*, and have nothing to do with chosing
audio *streams* (or tracks as you put it).​

So far, the behavior of -ac explained by others and by your link are the
same, and match the current behavior of ffplay I'm observing. What exactly
is your problem?
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Re: [FFmpeg-user] on frame changes in recode

[Cley Faye]

2016-04-28 23:31 GMT+02:00 Xen :

>
> All of this is unnecessary, but if it makes you happy, here you have it ;P.


​I was curious about your issue, and I too was eagerly waiting for that
log. Sadly it didn't come.
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Re: [FFmpeg-user] Android M crash when use ffmpeg, the log shows ffmpeg.so not found, only happen in Android M but not in other versions

[Cley Faye]

2016-04-18 11:25 GMT+02:00 Liu Ming :

> As the subject says, can you check it and reply as soon as possible,


​I might be mistaken, but I doubt you'll get much help here for the
following reasons:
- This list is mainly for the CLI tools
- You give no information about the ffmpeg version you're trying to use or
any form of full logs
- It seems that this will ultimately be a linker issue and/or an
android-specific build issue, and as far as I know, the common answer here
is "ffmpeg only provide source". Maybe the devel list is more appropriate.
- Also, the way to put the body of the message in the subject line can be
seen as rude by some people

All in all, I don't think this is the right place to ask this; maybe the
dev list, or the place that provided you your build.

Regards,
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Re: [FFmpeg-user] error

[Cley Faye]

2016-04-12 20:16 GMT+02:00 Ana :

> This is what i am getting


​You might be interested in the first line of that page:
https://ffmpeg.org/mailman/listinfo/ffmpeg-user

> Please provide your ffmpeg command(s) and complete, uncut, unscripted
console output(s) when possible.​

​Some things are missing from your messages, namely the ffmpeg command, and
the complete, uncut, unscripted console output.

Try posting your command, and redirect all output to a file so it can be
made available here. Every part printed might have a good clue about what's
going on, hence the request for full output.​
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Re: [FFmpeg-user] Syntax problem for specifying files(?)

[Cley Faye]

2016-04-06 18:28 GMT+02:00 aviv solodoch :

> I'm trying to create a movie from a series of .png files.
> The file names are of the form "NWAT_Vort_GB_.pg",
> where  is a 4 digit number. The lowest value is =1148.
>

​In addition to the other reply, I think this link can be useful:
https://ffmpeg.org/ffmpeg-formats.html#image2-1
There are a handful of options here that can help you, including obviously
start_number.
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Re: [FFmpeg-user] ffserver with libvpx-vp9 problem

[Cley Faye]

2016-03-31 11:24 GMT+02:00 Moritz Barsnick :

> a) Why are you using "-pass 1"? AFAIU it will still create an encoded
> video, but it is really meant for multi-pass encoding.
>
> b) What do you intend to achieve with "-b:v 0"? No video in the world
>can compress that well. ;-) (I'm not aware of any special semantics
>of the value '0' either.)
>

About that, I can enlighten you: "-b:v 0" ​is used for Constant Quality
encoding for vp9; it works with -crf to determine the "visual quality" of
the video. Setting both -b:v and -crf result in constrained quality (like
constant quality with a maximum bitrate value).

A good read on this: https://trac.ffmpeg.org/wiki/Encode/VP9
However, there is no mention of multiple passes, and it wouldn't do much
good in constant quality encoding anyway, as two-pass is used to improve
variable bitrate encoding (at least for other codecs).
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[FFmpeg-user] "Cannot allocate memory" error

[Cley Faye]

Hi,

I'm facing an unusual issue that seems caused by amix, but I'm not sure
about this at all.

I'm trying to merge the audio from a video file and an audio file, and
output a single video file with one video stream and one audio stream. I
stripped down the command to this:
> ffmpeg -i "in.mp4" -i "in.m4a" -filter_complex "[0:a][1:a]amix[outA]"
-map 0:0 -map [outA] out.mp4

(of course the command I intend to use have a lot more stuff going on, but
I get the error with just this command line).

And I get the following error message: "Cannot allocate memory"

The output:

ffmpeg version N-78990-g689211d Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype
--enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus
--enable-librtmp --enable-libschroedinger --enable-libsnappy
--enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg
--enable-lzma --enable-decklink --enable-zlib
  libavutil  55. 19.100 / 55. 19.100
  libavcodec 57. 28.100 / 57. 28.100
  libavformat57. 28.100 / 57. 28.100
  libavdevice57.  0.101 / 57.  0.101
  libavfilter 6. 39.102 /  6. 39.102
  libswscale  4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc54.  0.100 / 54.  0.100
[h264 @ 01e2e394c6a0] Increasing reorder buffer to 1
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'in.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
title   : LP 37
date: 2016-03-10
encoder : Lavf57.27.100
  Duration: 00:00:05.02, start: 0.00, bitrate: 20188 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv,
bt709), 1920x1080 [SAR 1:1 DAR 16:9], 19953 kb/s, 60 fps, 60 tbr, 60k tbn,
120 tbc (default)
Metadata:
  handler_name: VideoHandler
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, fltp, 221 kb/s (default)
Metadata:
  handler_name: SoundHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'in.m4a':
  Metadata:
major_brand : M4A
minor_version   : 512
compatible_brands: isomiso2
title   : LP 37
date: 2016-03-10
encoder : Lavf57.27.100
  Duration: 00:00:05.02, start: 0.021333, bitrate: 131 kb/s
Stream #1:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, fltp, 128 kb/s (default)
Metadata:
  handler_name: SoundHandler
[libx264 @ 01e2e43b1880] using SAR=1/1
[libx264 @ 01e2e43b1880] using cpu capabilities: MMX2 SSE2Fast SSSE3
SSE4.2 AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 01e2e43b1880] profile High, level 4.2
[libx264 @ 01e2e43b1880] 264 - core 148 r2665 a01e339 - H.264/MPEG-4
AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html -
options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12
lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0
bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'out.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
title   : LP 37
date: 2016-03-10
encoder : Lavf57.28.100
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021),
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 60 fps, 15360 tbn, 60 tbc
(default)
Metadata:
  handler_name: VideoHandler
  encoder : Lavc57.28.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz,
stereo, fltp, 128 kb/s (default)
Metadata:
  encoder : Lavc57.28.100 aac
Stream mapping:
  Stream #0:1 (aac) -> amix:input0 (graph 0)
  Stream #1:0 (aac) -> amix:input1 (graph 0)
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  amix (graph 0) -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
frame=   75 fps=0.0 q=31.0 size=  94kB time=00:00:00.30
bitrate=2557.8kbits/s