Re: [FFmpeg-user] Piped processes result in glitches/dropouts

[Moritz Barsnick via ffmpeg-user]

Hej Anders,

On Sat, Sep 14, 2024 at 13:32:01 +0200, Anders Bjerkén wrote:
> ffmpeg -f alsa -i hw:2 -filter:a "volume=1.5" -ab 192k -acodec mp3 -f mp3 -
> | ffmpeg -f mp3 -i - -c copy -f mpegts udp://:1685 -c copy -f
> mp3 udp://0.0.0.0:1685

It's usually expected to also post the complete, uncut console
output(s) here.

> Conceptually, it works. However, every now and then I get synchronized 1-2
> second long glitches in the outcoming data stream for both outputs.

This sounds like a buffering issue.

What happens if you stream directly from the first ffmpeg process to
one of the outputs? Does it also glitch?

You could consider using one ffmpeg process and the "tee" muxer. I'm
not sure it will solve your problem, but could be an alternative and a
more lightweight command line.

> have used all suggested versions of this command mentioned
> on https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs, but with the
> same result in the end.

Ah, I forgot that the "tee" muxer is mentioned there as well.

Moritz
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Re: [FFmpeg-user] Not able to configure ffmpeg 7.0.1 through minGW compiler

[Moritz Barsnick via ffmpeg-user]

On Thu, Aug 01, 2024 at 06:43:19 +, FFmpeg user discussions wrote:

> I am getting error saying "Compiler lacks C11 support" however I was
> not facing this issue when I used same compiler for ffmpeg 6.0

That's quite expected. The Changelog say so:
https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n7.0:/Changelog

Do you not have a newer compiler available?

https://en.wikipedia.org/wiki/C11_(C_standard_revision)#Compiler_support

Cheers,
Moritz
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Moritz Barsnick via ffmpeg-user]

On Fri, May 24, 2024 at 02:23:16 +0200, Ferdi Scholten wrote:
> > what is this supposed to be?
> > 16 Bit with 44,1 kHz (CD) are sufficient.
> > So, we have PCM_S16BE and PCM_S16LE.
> >
> > Whats the exact command to get it done?
>
> Youtube simply does not accept lossless audio in a video (at all). You have
> to encode according to their specifications. They explicitly say to use AAC
> in .mov containers, so do not use anything else as it will be not be
> accepted.

As far as I understand these help pages, this is a recommendation.
Indeed, YouTube also uses ffmpeg (so they say) for processing their
files, and accepts quite a lot. They may reject some formats and codecs
by filter - e.g. to avoid uncompressed input - but it's a matter of
trial and error and experience.

(Streaming to YouTube "live" is even more tricky.)

> You can only upload lossless audio (as .wav or .flac) as extra language for
> an existing uploaded video

Is that so?

BTW, the pages saying that PCM in WAV is accepted may either be
enormously outdated, or referring to YouTube Music. You really need to
be careful with trusting search results.

> this lossless audio will be converted to a lossy
> format by Youtube that will fit the videostream (usually AAC or Opus).

THIS is the important point. While providing lossless input in order to
not lose quality even before submission is nice idea (if it works), you
will never get lossless formats back when playing the media from
YouTube.

And lossless video is not really bandwidth-friendly anyway, but that
wasn't the question obviously.

Cheers,
Moritz
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Re: [FFmpeg-user] abitscope won't run

[Moritz Barsnick via ffmpeg-user]

On Mon, May 20, 2024 at 13:14:43 -0400, Mark Filipak wrote:
> Doesn't work:
> ffmpeg -i c:\source.mp4 -map 0 -af abitscope -c:v libx265 -x265-params
> crf=18 -c:a ac3 -sn -dn "c:\abitscope.mp4"
[...]
> [vf#0:0 @ 0053e0c0] Tried to connect video stream to audio 
> filtergraph input

abitscope has a video output. You need to use a "-filter_complex"
instead of "-af" to map its output to a video stream (and then of
course map that stream into your output file).

Cheers,
Moritz
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Re: [FFmpeg-user] VBR encoding not supported ???

[Moritz Barsnick via ffmpeg-user]

On Wed, May 08, 2024 at 12:09:52 +0530, Aditya Dandavate wrote:
> As I use FFMPEG mostly for creating videos that are uploaded to Youtube, is
> it fine to use VP8/Opus in a .webm container ?

As far as I understand, YouTube does support those formats and codecs
for uploads. Its recommendations are different though, but feel free to
ignore them, if it works for you:

https://support.google.com/youtube/answer/1722171?hl=en

> Also, Youtube does support VP8, right ?

Yes, AFAIK. Just try it. ;-)

Cheers,
Moritz
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Re: [FFmpeg-user] VBR encoding not supported ???

[Moritz Barsnick via ffmpeg-user]

On Mon, Apr 29, 2024 at 18:47:36 +0530, Aditya Dandavate wrote:
> Also, according to `Guidelines for high-quality lossy audio encoding`
> guide, is it true that only E-AC3 is "officially" supported in mp4 ? Does
> this mean that all other audio codecs like Opus, AAC, mp3 are
> illegally/unofficially supported by the container ?

There may be unofficial ones, but your list is not correct.

Wikipedia has a table and references to the ISO standards documents:
https://en.wikipedia.org/wiki/Comparison_of_video_container_formats#Audio_coding_formats_support

E-AC3 is certainly not the only officially supported audio codec.

Moritz
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Re: [FFmpeg-user] Extract sound from multiple input files

[Moritz Barsnick via ffmpeg-user]

On Mon, Apr 29, 2024 at 11:48:18 +0200, Michael Koch wrote:
> Am 29.04.2024 um 11:28 schrieb turgut kalfaoğlu:
> > Hi. I have an interesting case. I need to extract sound from multiple
> > input files,  combine those video files into one, and then re-insert the
> > audio into the combined video file.
> >
> > $ ffmpeg -i resized/360_0233.MP4   -i resized/360_0234.MP4   -i
> > resized/360_0235.MP4 -vn -map 0:a   tmpsound.mp3

You explictly selected only the fist input with "-map 0:a".

> Not tested, but I think it should work with the concat filter:
>
> ffmpeg -i resized/360_0233.MP4 -i resized/360_0234.MP4 -i
> resized/360_0235.MP4 -lavfi [0][1][2]concat=n=3:v=0:a=1 -y tmpsound.mp3

I have a different interpretation of what "combine those into one" was
supposed to mean. ;-)

Turgut, can you be more precise? The three audio tracks, what shall
happen with them?

* Attach audio one after the other on the timeline (i.e. concatenate).
* Insert all three audio tracks as separate parallel tracks into the
  resulting video.
* Overlay all three audio tracks, so that you hear their sound
  simultaneously in the final track.

Michael's solution is for problem no. 1.

Cheers,
Moritz
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Re: [FFmpeg-user] Best way convert to mpg

[Moritz Barsnick via ffmpeg-user]

On Wed, Jan 10, 2024 at 23:30:33 +0100, Reindl Harald wrote:
> Am 10.01.24 um 22:38 schrieb Budi Janto:
> > In last week I tried to convert *.dat, *.avi and other video format to
> > *.mpg.

> "-vcodec copy -acodec copy" - so you specify don't re-encode anyhting which
> can be done also by changing the file extension, both don't make any sense

I cannot speak for the *.dat files, but AVI containers aren't
compatible with MPEG containers.

Moritz
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Re: [FFmpeg-user] Can I add a codec for compatibility and automatically keep the original?

[Moritz Barsnick]

Hi,

On Sat, Nov 11, 2023 at 11:12:07 +0100, bbb wrote:
> when I have a file using advanced subtitles, I often want to keep those for
> most devices but also add a basic subtitle format that is supported by my
> TV.
>
> So far, I've been converting the subtitles and after that, mapping the
> different subtitle format from the original and the output file into a final
> file in a second operation. Surely it would also be possible to map the
> original subtitles twice and then copy all codecs and convert only that of
> the second subtitle stream.

Sure you can. You map the relevant input stream twice, and copy the
first one while converting the second one.

Something like:

$ ffmpeg -i video.ext -map 0:v -map 0:a -map 0:s -map 0:s -c:v copy -c:a copy 
-c:s:0 copy -c:s:1 srt output.ext

(Or perhaps convert the first one, if it's important for your TV.)

> I am planning to work on an automation (that works on a directory without
> knowing the number of streams in each file) and wondering, if ffmpeg maybe
> already has the kind of feature that I am looking for.

I believe ffmpeg cannot automate the work on the number of streams. For
the above scenario, you have to identify the streams first before
constructing the command line. (So there's no "-map_if_exists 0:s"...)

Cheers,
Moritz
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Re: [FFmpeg-user] Meaning of -color_range 2 ?

[Moritz Barsnick]

On Sun, Oct 01, 2023 at 14:05:38 +0200, Paul B Mahol wrote:
> >> I guess it stands for "-color_range limited", but even that I don't
> >> understand what it means.
> >> Where does the calculation "MPEG (219*2^(n-8))" come from?
> > And how can I know, in which version of FFMPEG this patch was released?
> >
> > https://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=7ca71b79f2b3256a0eef1a099b857ac9e4017e36
> >
>
> By searching for that commit log in git log of branch you use currently.
> By searching the source code of branch you use currently to have that
> added/changed lines from such commit log.
> By looking at date of commit and comparing to date of release of
> FFmpeg version currently used.

This also gives a quick overview:

$ git tag --contains 7ca71b79f2b3256a0eef1a099b857ac9e4017e36
n5.0
n5.0.1
n5.0.2
n5.0.3
n5.1
n5.1-dev
n5.1.1
n5.1.2
n5.1.3
n5.2-dev
n6.0
n6.1-dev

Means the first release it was integrated into was 5.0.

(When the patch is backported to old branches, it gets a different
hash, though. There, you really need to check manually. Usually, the
commit logs then say "cherry-picked from
7ca71b79f2b3256a0eef1a099b857ac9e4017e36".)

Moritz
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Re: [FFmpeg-user] Can transpose and between done in one command

[Moritz Barsnick]

Hi Cecil,

> In principle that would lead to a little better output video I think,
> because it is processed one time less. Or would it not make much
> difference?

Oh, yes, absolutely it would make a difference. Each (lossy) reencoding
is a loss in quality. Also, you could save a lot of time.

As the ffmpeg-filters docs say,
> Filters in the same linear chain are separated by commas, [...]

You're already doing this with "select,(a)setpts".

I would suggest selecting first, then transposing (which saves you from
transposing content which is later discarded), but it shouldn't matter.

I'm just not sure whether the t's in your select filter need to be
adapted, because you changed the video timestamps in the original
"first" step.

> -vf "transpose=cclock"  \

> -vf "
>  select='between(t,   0.7,  13.0) +
>  between(t,  26.0,  67.0) +
>  between(t,  82.0,  87.2)',
>   setpts=N/FRAME_RATE/TB
> "   \
> -af "
> aselect='between(t,   0.7,  13.0) +
>  between(t,  26.0,  67.0) +
>  between(t,  82.0,  87.2)',
>  asetpts=N/SR/TB
> "   \


So ultimately, your filters would be:

-vf "
 select='between(t,   0.7,  13.0) +
 between(t,  26.0,  67.0) +
 between(t,  82.0,  87.2)',
 setpts=N/FRAME_RATE/TB,
 transpose=cclock
"   \
-af "
aselect='between(t,   0.7,  13.0) +
 between(t,  26.0,  67.0) +
 between(t,  82.0,  87.2)',
 asetpts=N/SR/TB
"

I'm not sure how you manage to collect content from 82.0 to 87.2 in a
59 second long video, but I may be missing something. ;-)

Cheers,
Moritz
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Re: [FFmpeg-user] Can I change the image format from `showwavespic` to png?

[Moritz Barsnick]

Hi Marcin,

On Fri, Mar 10, 2023 at 06:04:15 +0100, Marcin Borkowski wrote:
> I use the `showwavespic` filter to get an image of the waveform.
> However, the default image format is jpg, and the only way I found I can
> change it is by providing a suitable extension of the output file.

Filters don't have codecs - they output raw frames, the rest of the
encoding chain determines the codec.

> Trouble is, I need the output to be stdout.

>From your previous thread, I'm guessing you're using "-f image2 -".

The help for the image2 format says:
Muxer image2 [image2 sequence]:
Common extensions: 
bmp,dpx,exr,jls,jpeg,jpg,jxl,ljpg,pam,pbm,pcx,pfm,pgm,pgmyuv,phm,png,ppm,sgi,tga,tif,tiff,jp2,j2c,j2k,xwd,sun,ras,rs,im1,im8,im24,sunras,vbn,xbm,xface,pix,y,avif,qoi.
Default video codec: mjpeg.

That's where the jpg codec comes from. So what you need to do is to
give an explicit image codec to use.

> Is it possible to output png to stdout?

All you need to do is to add "-c:v png" as an output option, i.e.
"-c:v png -f image2 - "

Cheers,
Moritz
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Re: [FFmpeg-user] Repeatable resolution issue exporting mxfs, 1080p is coming out as 1088p

[Moritz Barsnick]

On Mon, Feb 27, 2023 at 15:07:03 +, William Jones wrote:
> Unfortunately every time I try, the resulting file comes out as 1920x1088
> which breaks our pipeline. Ive tried using scale and have tried the source
> file being different formats (image sequences or .movs etc) and nothing has
> helped so far, which is why the below  code doesnt have any scale modifiers
> in it.
>
> the code i have been using is below:
>
> ffmpeg - start_number 1000 -i *-inputfile%4d.tiff* -c:v dnxhd -r 24
> -b:v115M -pix_fmt yuv422p *outputfile.mxf*

You'll have to show us the complete, uncut output of your ffmpeg command.

Also, which tool says it's 1088? Do you have mxfdump?

As far as I understand, MXF formats can have differing StoredHeight and
DisplayHeight, e.g. StoredHeight 1088, DisplayHeight 1080. (Not sure
whether that's what ffmpeg is doing.)

Moritz
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Re: [FFmpeg-user] Evade shortening audio while transcode

[Moritz Barsnick]

On Fri, Jan 20, 2023 at 14:59:20 +0200, Alexander Bieliaev via ffmpeg-user 
wrote:
> When transcoding audio from WAV and GSM formats to MP3, the result is
> shorter in duration than the original. How can you avoid this and keep the
> original duration?

Have you played the transcoded files, and is something actually
missing? Or is it just the length indicated by the media players?

The original files may have misleading headers regarding their lengths.

Cheers,
Moritz
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Re: [FFmpeg-user] Passing headers to encrypted hls stream in ffmpeg

[Moritz Barsnick]

On Thu, Jan 26, 2023 at 20:43:53 +0530, Nitin Singh wrote:
> I am using ffmpeg 5.1. I want to save encrypted aes-128 hls in which
> the decrypting key link is in m3u8 stream link. I am able to pass
> headers & get m3u8 file content using :
>
> ffmpeg  -headers "Hvar1: hvalue1" -headers "Hvar2: hvalue2" -headers
> "Hvar3: hvalue3" -i "http://media.link/mediafile.m3u8"; -t 30 -c copy
> output.mkv -loglevel debug
[...]
> But the problem is with getting remote key file i.e.
> https://media.key.link/live/media-10.key as it needs the same headers
> to get fetched and those same headers are not getting passed to remote
> key file url and I get this error :

I believe that's a bug or shortcoming of ffmpeg, as described in this
ticket:
https://trac.ffmpeg.org/ticket/7695

When searching for this, I didn't realize I had commented on this
almost exactly four years ago. ;-) I probably had the same issue.

Moritz
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Re: [FFmpeg-user] MKV to MP4 subtitle issue

[Moritz Barsnick]

On Tue, Jan 17, 2023 at 10:02:59 +0100, pehache wrote:
> > MP4 cannot carry those HDMV PGS subtitles (or any other bitmap based
> > format, if I remember correctly).
>
> MP4 can carry the DVDSUB format, which is bitmap.

Ah great, sorry, I wasn't aware (anymore)

> The OP can give a try transcoding the PGS subtitles to DVDSUB, using "-c:s
> dvdsub", but in my (limited) experience it often fails for some reason or
> another.

Indeed, I have always had problems converting any bitmap-based format
to dvdsub. I may dig into it again, but the original poster should
perhaps just try.

Moritz
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Re: [FFmpeg-user] MKV to MP4 subtitle issue

[Moritz Barsnick]

On Tue, Jan 10, 2023 at 19:23:33 -0600, Steve Zemlicka wrote:
> The question I have is similar to the following post:
> https://superuser.com/questions/932730/ffmpeg-mkv-to-mp4-conversion-loses-subtitles

Those answers are not helpful, because you have HDMV PGS, which is a
bitmap format, that cannot be converted to ASS or SRT directly.

> I am using ffmpeg version n5.1.2.  I have an mkv that has two subtitle
> tracks.  I am trying to convert it to mp4.
>
> mkvmerge -i input.mkv
> File 'input.mkv': container: Matroska
> Track ID 0: video (HEVC/H.265/MPEG-H)
> Track ID 1: audio (E-AC-3)
> Track ID 2: subtitles (HDMV PGS)
> Track ID 3: subtitles (HDMV PGS)
> Chapters: 10 entries
>
> "subtitles=input.mkv:si=1" as per
> http://ffmpeg.org/ffmpeg-filters.html#subtitles.  Unfortunately this
> results in an error:
> "Unable to find a suitable output format for 'subtitles=input.mkv:si1'"
> followed by:
> "subtitles=input.mkv:si=1: Invalid argument"

If you use the "subtitles" filter, you must declare it after the
leading filter option "-vf".

But, that said, the subtitles filter is actually meant for handling
text-based subtitles. Which it what you presume here:

> I've also tried several other options including "-c:s copy" which didn't
> work due to the nature of the original subtitles (PGS) I suspect.

MP4 cannot carry those HDMV PGS subtitles (or any other bitmap based
format, if I remember correctly).

Since you prefer hardsubs anyway, perhaps this method mentioned in
ffmpeg's own wiki can be of help:
https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo#Picture-basedsubtitles

BTW, here's a link on how to extract those bitmaps:
https://superuser.com/q/1748962
You would need OCR to convert them to text-based. ffmpeg has no
integration of such tools, but external tools are available. (I have
one myserlf attempted to convert bitmap subs to text-based subs. It was
a cumbersome process.)

Cheers,
Moritz
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Re: [FFmpeg-user] How to compile ffmpeg with https support

[Moritz Barsnick]

On Tue, Nov 15, 2022 at 09:34:56 +0800, Sendex Six wrote:
> Hi all, as per the title of this email, how can we do this? Also, please
> include links for any references. Thank you so much, everyone.

Check the output of "./configure --help":
  --enable-gnutls  enable gnutls, needed for https support
   if openssl, libtls or mbedtls is not used [no]
  --enable-libtls  enable LibreSSL (via libtls), needed for https 
support
   if openssl, gnutls or mbedtls is not used [no]
  --enable-mbedtls enable mbedTLS, needed for https support
   if openssl, gnutls or libtls is not used [no]
  --enable-openssl enable openssl, needed for https support
   if gnutls, libtls or mbedtls is not used [no]
  --disable-schannel   disable SChannel SSP, needed for TLS support on
   Windows if openssl and gnutls are not used 
[autodetect]

So, you need to provide one of these libraries and add the appropriate
configure option (except for Windows, where SChannel is automatically
detected).

Cheers,
Moritz
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Re: [FFmpeg-user] m3u8 download has no audio track

[Moritz Barsnick]

Hi Carl,

On Mon, Nov 07, 2022 at 21:01:38 -0500, Carl Fink wrote:
> ffmpeg -i"http://example.com/video_url.m3u8";  -c copy -bsf:a aac_adtstoasc 
> "output.mp4"
[...]
> This has worked in the past, but lately from Nebula, I get an output
> file with no audio track, according to ffplay, and also vlc and mplayer.

If you show us the complete output of the original command, we could
see that the stream probably doesn't contain the audio at all.

How do you obtain the stream URL?

> Suggestions? What information can I post that will help to debug this.

Have you tried using yt-dlp for downloading your content? It is usually
aware of separate stream URLs for video and audio, if that is the case.

> Note: I am a paying Nebula subscriber and do not pirate their content.
> I just want to be able to watch it offline, notably on planes.

I have absolutely no problem with that. :)

Cheers,
Moritz
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Re: [FFmpeg-user] How to add black bars as replacement for height?

[Moritz Barsnick]

On Tue, Nov 08, 2022 at 03:02:36 +0100, FFmpeg user discussions wrote:
> Can I use ffmpeg to convert a 1920x800 resolution clip to 1920x1080? The
> goal is to fill the remaining pixels with a true black bar.

Have a look at the pad filter:
https://ffmpeg.org/ffmpeg-filters.html#pad-1

Something like
$ ffmpeg -i inputfile -vf "pad=h=1080:y=(oh-ih)/2" outputfile

The "y" expression is for centering the video vertically.

Cheers,
Moritz
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Re: [FFmpeg-user] Image to Video conversion

[Moritz Barsnick]

On Mon, Sep 19, 2022 at 15:13:27 +0200, Naveen.B wrote:
> I tried to mux the images, with the below command, but I got an error
> "No bmp codec tag found for  codec bmp"
> ffmpeg -framerate 30 -i CapturedImage-001.bmp -codec copy output.mkv.
> I would appreciate it if you could suggest a command for converting
> .bmp to lossless video.

You can't put BMP-encoded images directly into a video. You'll need to
(losslessly) re-encode to some other codec, such as FFV1 or VP9
lossless, or whatever your target device/audience may be willing to
support.

Cheers,
Moritz
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Re: [FFmpeg-user] Need to Convert MP4 (or MKV) with AAC *and* AC-3 Audio to Multi-Rendition HLS (each with *both* audio streams)

[Moritz Barsnick]

Hi Clay,

On Tue, Aug 30, 2022 at 18:51:13 -0400, Clay wrote:
> If you have a better way to achieve this, I am ***all ears***.
> Here is the code I have so far:

I think you need to fix your command line in various places:

> ffmpeg -i ../InputFile-1920x804-24f.mp4 \
> -filter_complex "[0:v]split=4[v1][v2][v3][v4]; [v1]copy[v1out];
> [v2]scale=w=1280:h=536[v2out]; [v3]scale=w=720:h=300[v3out];
> [v4]scale=w=480:h=200[v4out]"\
> -map [v1out] -c:v:0 libx264 -x264-params -b:v:0 -maxrate:v:0 4M

"-x264-params" needs an argument - that's missing here.
"-b:v:0" needs an argument - that's missing here.

And for bash's sake, I believe you need to quote "[v1out]" and the
similar ones below (but I may be overambitious here).

> -minrate:v:0 4M -bufsize:v:0 4M -crf 17 -preset slower -g 48
> -sc_threshold 0 -keyint_min 48 \
> -map [v2out] -c:v:1 libx264 -x264-params -b:v:1 -maxrate:v:1 3M
> -minrate:v:1 3M -bufsize:v:1 3M -crf 17 -preset slower -g 48
> -sc_threshold 0 -keyint_min 48 \
> -map [v3out] -c:v:2 libx264 -x264-params -b:v:2 -maxrate:v:2 2M
> -minrate:v:2 2M -bufsize:v:2 2M -crf 17 -preset slower -g 48
> -sc_threshold 0 -keyint_min 48 \
> -map [v4out] -c:v:3 libx264 -x264-params -b:v:3 -maxrate:v:3 1M
> -minrate:v:3 1M -bufsize:v:3 1M -crf 17 -preset slower -g 48
> -sc_threshold 0 -keyint_min 48 \
> -map 0:a -c:a:0 \
> -map 0:a -c:a:1 \
> -map 0:a -c:a:2 \
> -map 0:a -c:a:3 \

"-c:a:x" needs an argument.

> -f hls \
> -hls_time 2 \
> -hls_playlist_type vod \
> -hls_flags independent_segments \
> -hls_segment_type mpegts \
> -hls_segment_filename OutputFile_%v/data%02d.ts \
> -master_pl_name master.m3u8 \
> -var_stream_map "v:0,a:0 v:1,a:1 v:2,a:2 v:3,a:3"OutputFile_%v.m3u8

> This code breaks when I attempt to run it, giving the following error:
> [NULL @ 0x55f0357bbd40] Unable to find a suitable output format for '0:a'
> 0:a: Invalid argument

Probably because ffmpeg's whole option parsing is messed up due to your
missing arguments.

> I have tried many varieties of the audio stream mapping... 0:a, a:0,
> a:0:1.. etc.

And this is not the actual issue.

Cheers,
Moritz
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Re: [FFmpeg-user] How to embed subtitles into video using Python

[Moritz Barsnick]

On Wed, Jul 06, 2022 at 13:36:30 -0700, Jim DeLaHunt wrote:
> On 2022-07-06 02:46, Pavel Yermolenko via ffmpeg-user wrote:
> > In Python there is ffmeg support package: *ffmpeg-python*. ffmpeg-python
> > 0.2.0 
> > Unfortunately in the documentation [there] is no examples on how to
> > process subtitles.

> So I guess that this asks you to do either (and stand by for a discussion of
> the really big problem with this):
>
> ffmpeg
>     .input(input_dir+video_file)
>     .input(input_dir+subtl_file)
>     .get_args(map='0:v', map='0:a', c='copy', map='1', 'c:s:0'='mov_text', 
> 'metadata:s:s:0'='language=fra') ***Big problem!
>     .output(input_dir+'output.mp4')
>     .run()

In their test code, there is an "example" for maps:
https://github.com/kkroening/ffmpeg-python/blob/6189cd6861a90f6f52e6a8ba2db0fada54134194/ffmpeg/tests/test_ffmpeg.py#L657

> args = ffmpeg.output(i1['1'], i1['2'], TEST_OUTPUT_FILE1).get_args()
for "-map 0:1 -map 0:2".

You can probably reverse-engineer that, if the documentation doesn't
suffice.

Cheers,
Moritz
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Re: [FFmpeg-user] Need Command

[Moritz Barsnick]

On Thu, Jun 30, 2022 at 19:40:58 +0530, Dhwani Shukla wrote:
> I need command for merging Image, audio and gif into video for iOS swift.
> Currently I’m using following command which generating video but gif not 
> working :

Please elaborate on "not working".

> "-ignore_loop 0 -i \(objCreate.dwGIFPath ?? "") -loop 1 -i \(imagePath ?? "") 
> -i \(objCreate.dwAudioPath ?? "")  -vf 'scale=trunc(iw/2)*2:trunc(ih/2)*2' 
> -shortest -strict -2 -c:v libx264 -threads 4 -c:a aac -b:a 192k -pix_fmt 
> yuv420p -shortest \(myDocumentPath1)”

Please show us the actual resulting, complete command line, along with
the complete console output.

Moritz
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Re: [FFmpeg-user] Vertically flipped mp4 video

[Moritz Barsnick]

On Tue, Jun 28, 2022 at 14:55:58 +0200, Ulf Zibis wrote:
> Am 28.06.22 um 13:45 schrieb Ulf Zibis:
> > how can I correct a vertically flipped mp4 video without re-encoding? The 
> > left side is on the right side and vice versa.
> I tried:
> ffmpeg -i input.mp4  -c copy -metadata:s:v:0 flip=horizontal output.mp4
> But this changed nothing :-(

Metadata flags do exist for rotating, but not for flipping. Thus this
cannot be done without re-encoding.

Even for rotation, the player needs to understand the flag. This is not
the case for all players.

Cheers,
Moritz
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Re: [FFmpeg-user] Pictures in video

[Moritz Barsnick]

On Tue, May 24, 2022 at 21:55:20 +0200, FFmpeg user discussions wrote:
> What I find strange is that the input video is 23 MB and the picture
> is 302 KB, but the generated video is 42 MB.
> Am I doing something wrong?

No, that's not strange at all. You are reencoding a video. That process
will likely use different parameters than the original encoding.
(Approximately the same would happen without the overlay.) You can
control the encoding parameters, e.g. by the bitrate option "-b:v", or
with "-crf" (if using libx264 or libx265). There is a lot of
information about that available.

Moritz
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Re: [FFmpeg-user] FFMpeg - How to get the filesize in advance after encoding

[Moritz Barsnick]

On Tue, May 24, 2022 at 15:50:27 +0800, Sendex Six wrote:
> Thanks for your response,
>
> What is the formula, logic, or how to compute the file size from a video or
> audio file? after encoding?
> so that in the webpage we can set the content-length of video or audio

Only under very particular circumstances you can determine the file
size beforehand. I would say these are:
- raw formats/codecs with coonstant bitrates
- some containers with constant muxrates.

And in addition, it depends on the codec and the container.
Uncompressed WAV e.g. is extremely predictable. If you manage to get a
constant bitrate encoding and muximg, you might be able to determine
the size of MP4 or MPEG-TS, but you would have to know about the muxing
mechanisms and the overhead, and ffmpeg's settings influencing it.

As Harald said, the Content-Length isn't necessary, but the receiving
end (user/web browser) will not show the progress if it's missing. This
may or may not be okay.

For what you are trying to achieve, segmented streaming formats such as
HLS were made for: You can segment on the fly, and each finalized
segment's size is known to the receiving end. I guess you can also
indicate an average bitrate and a duration beforehand, but in my
experience it doesn't have to be corrent.

Cheers,
Moritz
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Re: [FFmpeg-user] issues in ffmpeg installation

[Moritz Barsnick]

On Fri, Apr 22, 2022 at 05:37:46 +, FFmpeg user discussions wrote:
> ./configure --enable-cuda --enable-cuvid --enable-nvdec --enable-nvenc 
> --enable-nonfree --enable-libnpp --extra-cflags=-I/usr/local/cuda/include 
> --extra-ldflags=-L/usr/local/cuda/lib64 --disable-x86asm
>
> below is the error i have got ,can u please help us in fixing the issue.i 
> have attached the config file log for reference as well
> [root@localhost FFmpeg]# ./configure --enable-nonfree --enable-cuda-nvcc 
> --enable-libnpp --extra-cflags=-I/usr/local/cuda/include 
> --extra-ldflags=-L/usr/local/cuda/lib64 --disable-static --enable-shared 
> --disable-x86asm
> ERROR: failed checking for nvcc.

Your log file says:
> nvcc -gencode arch=compute_35,code=sm_35 -O2 -std=c++11 -m64 -ptx -c -o 
> /tmp/ffconf.jxP0ew2J/test.o /tmp/ffconf.jxP0ew2J/test.cu
> ./configure: line 991: nvcc: command not found
> ERROR: failed checking for nvcc.

Did you install the nvidia-cuda-toolkit and nv-codec-headers?

Moritz
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Re: [FFmpeg-user] custom codec

[Moritz Barsnick]

On Tue, Apr 19, 2022 at 20:28:57 -0700, Right On wrote:
> hi - i was wanting to encode a video using the amv4
>  codec. this codec is currently
> not supported in ffmpeg - is there a way to add it? it's installed to my
> computer so i assumed there would be a way to call it but it doesn't come
> up when i run ffmpeg - encoders

No, ffmpeg cannot use arbitrary DLL codecs.

If I remember correctly, mencoder was able to do this, but possibly
only for a set of known codec DLLs.

Moritz
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Re: [FFmpeg-user] How to extract all the pictures but no repeat in an mp4 video file?

[Moritz Barsnick]

On Thu, Apr 21, 2022 at 22:28:36 +0800, mikelee164396 wrote:
> I want to extract all the pictures in an mp4 video file , but don't repeat 
> it. How to implement it with ffmpeg?

For example:
$ ffmpeg -i inputvideo -vsync passthrough outputimage.%05d.jpg

Cheers,
Moritz
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Re: [FFmpeg-user] API documentation for FFmpeg 5.0

[Moritz Barsnick]

On Mon, Feb 28, 2022 at 08:29:35 +0100, Andreas Unterweger wrote:
> I wanted to follow up and this and ask if there is *any* documentation
> about how to build the (HTML) documentation for a particular version
> of the FFmpeg libraries.

It is part of the regular build process. If you do
./configure; make

the HTML docs are automatically built (see folder doc/). They probably
depend on some tools, presumably makeinfo.

> It would also be helpful if there were any pointers to how the
> nightly documentation is built exactly ([2] in the message above).

Well, either you grab a release for that release's docs, or you grab
nightly or lastest Git, to get the latest docs. They are part of the
source tree, so they always correlate.

Cheers,
Moritz
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Re: [FFmpeg-user] configure gives "ERROR: libzmq >= 4.2.1 not found using pkg-config"

[Moritz Barsnick]

On Wed, Feb 23, 2022 at 22:41:46 +0100, Reindl Harald wrote:
> > https://packages.debian.org/bullseye/arm64/libzmq3-dev/filelist
>
> and you think "libzmq3-dev" is >= 4.2.1?
> why?
>
> "ERROR: libzmq >= 4.2.1 not found using pkg-config"

Because that is what its pkg-config file claims:

-
$ ar -p libzmq3-dev_4.3.4-1_arm64.deb data.tar.xz | tar xOJvf - \*libzmq.pc
./usr/lib/aarch64-linux-gnu/pkgconfig/libzmq.pc
prefix=/usr
exec_prefix=${prefix}
libdir=${prefix}/lib/aarch64-linux-gnu
includedir=${prefix}/include

Name: libzmq
Description: 0MQ c++ library
Version: 4.3.4
Libs: -L${libdir} -lzmq
Libs.private: -lstdc++  -lpthread -lrt
Requires.private:  krb5-gssapi libsodium openpgm-5.3 >= 5.3 norm libbsd
Cflags: -I${includedir}
-

Name is libzmq, Version is 4.3.4. That should be >= 4.2.1.

I assume something else is wrong, like the pkg-config path not being
evaluated.

The output of
$ pkg-config --modversion libzmq
would be interesting, but presumably libzmq isn't found. I wonder how
to debug remotely what pkg-config is doing, if that dev package is
"correctly" installed.

Moritz
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Re: [FFmpeg-user] configure gives "ERROR: libzmq >= 4.2.1 not found using pkg-config"

[Moritz Barsnick]

On Wed, Feb 23, 2022 at 19:13:18 +0100, Moritz Barsnick wrote:
> Thomas, you will need to should us [...]

show us, I meant.

Moritz
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Re: [FFmpeg-user] configure gives "ERROR: libzmq >= 4.2.1 not found using pkg-config"

[Moritz Barsnick]

On Wed, Feb 23, 2022 at 17:16:43 +0100, Reindl Harald wrote:
> Am 23.02.22 um 12:24 schrieb Thomas Seilund via ffmpeg-user:
> > libzmq3-dev/stable,now 4.3.4-1 arm64 [installed]
> > libzmq5/stable,now 4.3.4-1 arm64 [installed,automatic]
>
> libzmq5 is *not* the dev-package

But the quoted libzmq3-dev is. It's apparently installed, and it should
include the required pkg-config file:

https://packages.debian.org/bullseye/arm64/libzmq3-dev/filelist

Thomas, you will need to should us the relevant section of ffbuild/config.log.
That section begins something like this:

> require_pkg_config libzmq libzmq >= 4.2.1 zmq.h zmq_ctx_new
> check_pkg_config libzmq libzmq >= 4.2.1 zmq.h zmq_ctx_new
> test_pkg_config libzmq libzmq >= 4.2.1 zmq.h zmq_ctx_new
> pkg-config --exists --print-errors libzmq >= 4.2.1
> check_func_headers zmq.h zmq_ctx_new -I/usr/include/pgm-5.2 
> -I/usr/lib/pgm-5.2/include -lzmq

Cheers,
Moritz
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Re: [FFmpeg-user] ERROR: libvmaf >= 2.0.0 not found using pkg-config

[Moritz Barsnick]

On Thu, Feb 03, 2022 at 21:07:08 +0900, 桃源老師 wrote:
> I experienced funny ffmpeg compille error "ERROR: libvmaf >= 2.0.0 not found 
> using pkg-config".  The reason why I feel funny is that ffmpeg commit of 
> "06a54a4ec73ea6eb8cec41b7af75367bafb075e0(Date: Sat Jan 22 10:32:34 2022 
> +0100)" can configure and make correctly, but current git head can't 
> configure.  I suppose something wrong was happened after 2022/01/22.
>
> I'm trying to build ffmpeg on macOS 12.2.

So, what's your actual version of libvmaf? Because _after_ commit
06a54a4ec73ea6eb8cec41b7af75367bafb075e0, the requirement was increased
to 2.0.0.

If you're using git anyway, you could do a "git bisect".

> /var/folders/1d/qk_hczkn68xfv_7gdbkmhwx4gn/T//ffconf.DmkkrFIx/test.c:3:44:
>  error: use of undeclared identifier 'vmaf_init'
> long check_vmaf_init(void) { return (long) vmaf_init; }

It seems to me that vmaf_init() had existed before, so I'm not sure why
this would happen.

Cheers,
Moritz
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Re: [FFmpeg-user] Patch fails to apply

[Moritz Barsnick]

Hi Bryce,

On Mon, Jan 31, 2022 at 23:14:50 +, Bryce Newman wrote:
> I am not sure why this patch is failing.
> https://patchwork.ffmpeg.org/project/ffmpeg/patch/93113e7f-9b7a-4423-9ad2-47f3af940...@gettyimages.com/

You should be asking on ffmpeg-devel. But anyway:

> Can someone please advise on how to resolve “error: corrupt patch at line 10“?

If you extract the patch from your email and try to apply, you might
see the error yourself. I think your mailer has corrupted the patch.
The function's opening bracket '{' shouldn't be on the first column in
the patch.

Did you attach the patch as an attachment?

It sending it with your mailer doesn't work on the list, you may want
to use "git send-email" instead (which is a bit tricky to control).

Cheers,
Moritz
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Re: [FFmpeg-user] Feeding frames to ffmpeg directly

[Moritz Barsnick]

Hi Violet,

On Sun, Jan 16, 2022 at 15:14:47 -0600, Violet White wrote:
> Hi, I have a program that generates frames of animation and saves
> them to individual files on a disk, but what I would like to do is avoid
> writing the images to disk and simply pipe them in.
>
> Can I expect this to just work if I write multiple png files to the same 
> stream?
> Or will I need to adjust my program's output format?

This should just work. ffmpeg has an input format called "png_pipe"
(and similar for other formats.

Assuming all images have the same dimensions and colorspace - though
I'm not sure that is a must - you can just do:

$ cat *.png | ffmpeg -f png_pipe -i - ...

Of course, you have no files on disk, as in this example. So you would
just change your program to output the consecutive PNGs to stdout, and
pipe that to the ffmpeg command.

$ ./your_program | ffmpeg -f png_pipe -i - ...

Cheers,
Moritz
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Re: [FFmpeg-user] Can MJPEG be live streamed without transcoding?

[Moritz Barsnick]

On Sun, Jan 09, 2022 at 09:59:24 -0500, John Van Ostrand wrote:
> I wonder if you could use netcat to direct the stream from Stdout to a
> network socket then use netcat at the other end to put it into the Stdin of
> the remote ffmpeg.

You don't need netcat, ffmpeg has TCP, UDP, HTTP (and further)
streaming options built in.

The "nut" container is a good choice for streaming for ffmpeg to
ffmpeg, it can handle quite a lot of codecs.

Moritz
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Re: [FFmpeg-user] Convert to XAVC

[Moritz Barsnick]

On Thu, Jan 06, 2022 at 16:28:26 -0800, Hanna Frangiyyeh wrote:
> Any idea what filter is to be used to convert to XAVC mxf format.

As far as I can tell, the x264 encoder has an xAVC-compatible mode.
Check the x264 option "avcintra-flavor" with the value "sony".

Cheers,
Moritz
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Re: [FFmpeg-user] Filter Question

[Moritz Barsnick]

Hi Hanna,

On Tue, Jan 04, 2022 at 14:26:29 -0800, Hanna Frangiyyeh wrote:
>
> PS D:\TEMP\Source Files\Audio>
> D:\Apps\AudioRendering\AudioRendering\AudioRendering\ffmpeg.exe -i
> "D:\TEMP\Source Files\
> 720E\MLB21_INT_PLAYER_720.mov" -i "D:\TEMP\Source
> Files\Audio\file_example_WAV_1MG.wav" -filter_complex 'split [rgb_in][
> alpha_in];[rgb_in] lutrgb=a=minval [rgb_out];[alpha_in] format=rgba, split
> [T1], fifo, lutrgb=r=maxval:g=maxval:b=maxval
> , [T2] overlay [rgb_out][T1] fifo, lutrgb=r=minval:g=minval:b=minval
 
There's still something fishy here. Should there also be a semicolon?

> [T2][alpha_out]' -map '[rgb_out]' "D:\TEMP\Source F
> iles\Done\MLB21_INT_PLAYER_720_ProRes422-1080_ST_F.MOV" -map '[alpha_out]'
> "D:\TEMP\Source Files\Done\MLB21_INT_PLAYER_7
> 20_ProRes422-1080_ST_M.MOV" -acodec copy -map 0:0 -map 1:a:0 -y  1:a:0 -y
[...]
> [AVFilterGraph @ 01c0146020c0] No output pad can be associated to link
> label 'T1'.

This message said you should be looking for something related to T1.

Cheers,
Moritz
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Re: [FFmpeg-user] ffmpeg 4.4.1 security issue

[Moritz Barsnick]

On Thu, Jan 06, 2022 at 13:12:51 +, FFmpeg user discussions wrote:
> I am currently a data scientist at USAA. I was trying to use FFMPEG 4.4.1 to 
> convert spex audio files to wav audio format.
>
> My security team denied the download of the package, and here is the 
> following explanation that they gave:
> DOWNLOAD DENIED: Muliple known vulnerabilities like CVE-2021-38171
> I was wondering how I can get this fixed or if it is already fixed in a later 
> version?

The fix is mentioned in the CVE 
(https://nvd.nist.gov/vuln/detail/CVE-2021-38171):

https://github.com/FFmpeg/FFmpeg/commit/9ffa49496d1aae4cbbb387aac28a9e061a6ab0a6

It was ported to the 4.4 branch here:

https://github.com/FFmpeg/FFmpeg/commit/fb993619d1035fa9646506925ea70fb122038999

and that is contained in release 4.4.1, as far as I can tell (by "git tag 
--contains
fb993619d1035fa9646506925ea70fb122038999").

So the CVE refers to version 4.4, and version 4.4.1 fixes this and is
therefore not affected, AFAICT.

You'll have to have your security team check 4.4.1. You may need to
check each CVE separately (they mention "multiple known
vulnerabilities"). If in doubt, disable the affected feature (as in
this case: the ADTS muxer).

Hope this helps,
Moritz
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Re: [FFmpeg-user] Getting m3u8 URL from SVTPlay channels?

[Moritz Barsnick]

On Sun, Dec 26, 2021 at 21:10:05 +0100, Moritz Barsnick wrote:
> I double-checked this. The svtplay-dl binary offered as a download
> (version 4.10) seems to be broken. It has a date parsing error which
> seems to be fixed in the git version (and the fix should be in 4.10, so
> d'uh).

Wrong - my mistake. 4.10 works just fine with my instructions.

Moritz
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Re: [FFmpeg-user] Getting m3u8 URL from SVTPlay channels?

[Moritz Barsnick]

On Sun, Dec 26, 2021 at 08:26:34 +0100, Bo Berglund wrote:
> 1) I already have several of the tools mentioned (ytube-dl, yt-dlp, svtplay-dl
> etc), but they did not manage to download the real-time streams from SVT.

I double-checked this. The svtplay-dl binary offered as a download
(version 4.10) seems to be broken. It has a date parsing error which
seems to be fixed in the git version (and the fix should be in 4.10, so
d'uh).

With the git version, I get this result:

[barsnick@goodtimes svtplay-dl]$ ./svtplay-dl --live --only-video -g 
https://www.svtplay.se/kanaler/svt1
https://svt1-d.akamaized.net/se/svt1/manifest.mpd?defaultSubLang=1

And ffmpeg can handle that given URL just fine.

(Funny thing: Since SVT seems to have separate substream manifests for
video and audio now, svtplay-dl apparently currently cannot give you
both those URLs with "-g", you get only a pure video URL. With the
"--only-video" only "--only-audio" option as shown above, I get the
overall manifest which includes three stream qualities. If you throw
that at ffmpeg, it will select the best stream qualities by itself, or
you can just do "-map 0:M -map 0:N".)

> 3) The problem is that some broadcasts are not available after the fact (like
> some sports events) and have to be "recorded" in real time off of the stream
> itself.

I agree, I do the same once in a while. :-)

> This is why I wanted to create a PVR script to handle that. The script would 
> be
> called with a start time, duration and channel name (SVT has 4 different
> channels) as well as an output file name. The scheduling would use the at
> command to start the script at a specific time for unattended recording.

I do this not with a start time, but as an "at" job.
$ at 21:55
> ffmpeg -i > 
> "https://svt1-d.akamaized.net/se/svt1/manifest.mpd?defaultSubLang=1"; -c copy 
> -t 2:10:00 /path/to/mounted/PVR/'SVT1 - The name of the show'.ts -y
> ^D

Lykke til,
Moritz
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Re: [FFmpeg-user] Getting m3u8 URL from SVTPlay channels?

[Moritz Barsnick]

On Sat, Dec 25, 2021 at 12:19:35 +0100, Reino Wijnsma wrote:
> On 2021-12-25T10:13:14+0100, Bo Berglund  wrote:
> > But on the following pages it does not work at all:
> > https://www.svtplay.se/kanaler/svt1?start=auto

As I had mentioned in the thread previously, it's not given that the
stream URL is embedded in the actual web page. It might be listed in a
JSON file, which might even be referenced from the web page via
JavaScript or similar.

> It's really bad practise to parse HTML with RegEx! It's as if you're using a 
> sledgehammer when you should've been using a pair of tweezers for instance.

For fishing a single URL from a well-known syntactical construct, it
might be "good enough".

> This div is where the magic happens. Take a good look at the
> browser's network-traffic. You're probably going to need the value of
> the "class"-attribute.

And this is what tools like youtube-dl (or its successor yt-dlp) and
svtplay-d do: They know well how to parse these pages, and can extract
the URL for you (check their "-g" options). You don't need to reinvent
the wheel. Unless you need to avoid python. ;)

https://github.com/yt-dlp/yt-dlp/
https://svtplay-dl.se/

(Jeg har faktisk VPN, og jeg har gjort akkurat dette før, med SVT og
lignende.)

Cheers,
Moritz
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Re: [FFmpeg-user] About the log4j vulnerability (CVE-2021-44228)

[Moritz Barsnick]

On Thu, Dec 23, 2021 at 21:26:48 +0900, Tomonari Katsumata wrote:
> -Is it affected by the log4j vulnerability (CVE-2021-44228) while using 
> ffmpeg3.4.6?

No.

Cheers,
Moritz
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Re: [FFmpeg-user] [EXTERNAL] Re: GitHub and Releases

[Moritz Barsnick]

On Wed, Dec 15, 2021 at 19:21:42 +, Bryce Newman wrote:
> So just talking about the patch now…
> Which branch should I be in to add my patch? I assume that is release/4.4?
> Here are the branches I see.

Any contributed patches should be developed against and apply to
master:
https://www.ffmpeg.org/developer.html

Often, they also apply cleanly to older branches. Yet that is only of
concern to you if you want to compile an older version with your patch.

If a contributed patch is eventually merged, it can be backported to
the old release branches, if it's considered a reasonable bug fix. This
backporting is then usually done by the upstream developers.

Cheers,
Moritz
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Re: [FFmpeg-user] [EXTERNAL] Re: GitHub and Releases

[Moritz Barsnick]

On Wed, Dec 15, 2021 at 16:44:13 +, Bryce Newman wrote:

> I am waiting on a patch
> https://patchwork.ffmpeg.org/project/ffmpeg/patch/dm6pr08mb60732b5d77656c74c8491c4f85...@dm6pr08mb6073.namprd08.prod.outlook.com/
> too be approved and it may not be approved/merged into 4.4.2, etc
> anytime soon,

I don't see that this is likely to be merged (i.e. backported) into
4.4.2. New features only go into the development branch, which will
then become 4.5 or whatever is next there.

> so I need to compile against what is a known working version of
> ffmpeg and external libs used in
> https://github.com/jrottenberg/ffmpeg/blob/main/docker-images/4.4/ubuntu2004/Dockerfile.

If ffmpeg 4.4.1 compiles againt those libs, so will ffmpeg git master
HEAD.

You can probably just replace
https://ffmpeg.org/releases/ffmpeg-${FFMPEG_VERSION}.tar.bz2
with a link to the snapshot:
https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2

And even apply your patch right after unpacking this.

Cheers,
Moritz
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Re: [FFmpeg-user] h.265 crf values corresponding with h.264

[Moritz Barsnick]

Hi Cecil,

On Wed, Dec 15, 2021 at 19:06:36 +0100, FFmpeg user discussions wrote:
> Does YouTube has a preference for one, or the other?

For accepting, or for preseting?

If you're uploading, YouTube will accept quite a lot. ;-)

For streaming, I see most docs recommend H.264, but I understand
they also accept HEVC and VP9. VP9 is probably even slower to encode
though.

For content delivery, YouTube prefers VP9.

Cheers,
Moritz
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Re: [FFmpeg-user] GitHub and Releases

[Moritz Barsnick]

Hi Bryce

On Wed, Dec 15, 2021 at 14:51:40 +, Bryce Newman wrote:

> I am trying to clone from GitHub(https://git.ffmpeg.org/ffmpeg.git)
> the same source code that can be found in
> https://ffmpeg.org/releases/ffmpeg-4.4.1.tar.bz2. What branch in
> https://git.ffmpeg.org/ffmpeg.git contains the same 4.4.1 source code
> as in https://ffmpeg.org/releases/ffmpeg-4.4.1.tar.bz2 When I
> checkout release/4.4 the source code from GitHub is different from
> the source in https://ffmpeg.org/releases/ffmpeg-4.4.1.tar.bz2.

Releases correspond to tags, not branches. You want to check out
tag "n4.4.1".

Well, "want". Why do you want the exact same source code?

(The 4.4 branch is for developing what becomes 4.4.2, 4.4.3, ...)

Cheers,
Moritz
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Re: [FFmpeg-user] "ffplay -report" creates text files with mixed line endings

[Moritz Barsnick]

Hi, Jaime,

On Tue, Nov 09, 2021 at 11:51:18 +, Jaime wrote:
> AV_LOG_FORCE_NOCOLOR=1 ffplay -report -rtsp_transport tcp rtsp://camera.lan
> Mailman is showing the report file as being of type: 
> "application/octet-stream":

Is this the actual file you created, or the one you fetched from
mailman?

> A non-text attachment was scrubbed...
> Name: ffplay-20211109-100211.log
> Type: application/octet-stream
> Size: 32011 bytes
>
> I've just checked the line endings using "dos2unix -ih":
>
> $ dos2unix -ih ffplay-20211109-100211.log
>  DOSUNIX MAC  BOM   TXTBIN  FILE
>   17 163 301  no_bomtextffplay-20211109-100211.log
>
> Interesting. Is this "by design", or a bug (or neither)?

If you mean ffmpeg:

I'm pretty sure that part of this is by design. You should check the
actual line endings. ffmpeg uses a pure "CR" (carriage return, often
displayed as '^M') to cyclically overwrite the status line, and also
seems to dump this into the report file. Those would be the "MAC"
endings, I guess, and your mail agent seems to identify the file as
binary, and therefore send it as "application/octet-stream".

I cannot explain the DOS and UNIX discrepancy though. I get pure Unix:

$ dos2unix -ih ffplay-20211109-135854.log
 DOSUNIX MAC  BOM   TXTBIN  FILE
   0  83 327  no_bomtextffplay-20211109-135854.log

On the other hand, if this was not your local file, it could be
mailman's fault, or by one of the mail servers.

We would need to see your actual file to understand this. Note, I have
mailed myself such a report file, and had the local mail agent(s)
changing it! I noticed the line endings changed when receiving through
Exchange/Outlook, but I don't know who the culprit was.

Again, the "MAC" line endings will probably make the attachment binary
nevertheless(?). You should check whether you can force the MIME type
when sending, if the problem is on your sending side, and not when
receiving though mailman.

Cheers,
Moritz
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Re: [FFmpeg-user] Dynamic range compression fail with: invalid argument

[Moritz Barsnick]

On Sat, Oct 23, 2021 at 16:17:02 +0200, Android PowerUser wrote:
> Paul B Mahol  schrieb am Sa., 23. Okt. 2021, 15:45:
> > Where is "-af "  part?
>
> But is not in any documentation.  I have no idea what that is for.

It's documented in the same place where the "compand" filter is
documented, in the general part:

https://ffmpeg.org/ffmpeg-filters.html#Filtergraph-syntax-1

Cheers,
Moritz
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Re: [FFmpeg-user] High audio latency (although low with ffplay!)

[Moritz Barsnick]

Hi Arif,

On Sun, Sep 26, 2021 at 14:21:43 +0200, Arif Driessen wrote:
> However, My inroad, as I see it, is essentially going from high  level to
> low level so probably somewhere in this order of contribution:
>
> 1) Documentation fixes / maybe other superficial maintenance tasks that
> need doing
> 2) Audio filters (because I understand music production, and a tiny bit of
> the math and higher-level DSP concepts
> 3) Pulse/Pipewire (as I'm on GNU/Linux)
> 4) ALSA mentioned afterwards as it's more lower-level
> 5) branching out to video
> 5) Migrating ffmpeg to Rust (hahaha joking)

This is exactly how I started off.

I recommend:
- Read https://ffmpeg.org/developer.html, especially
   https://ffmpeg.org/developer.html#Contributing
- Subscribe to the ffmpeg-devel mailing list.
- Observe that mailing for some time.
  - Understand what is expected from a contribution, and what the
pitfalls are. (Bad commit messages, incomplete context, bad code,
bad coding style, malformed patches, answering to the wrong thread,
top-posting, ...).
  - Understand how the review process goes.
- Pick your favorite nit. I started off with documentation buglets
  (spelling errors, missing documentation on options) and small code
  fixes.
- Also check trac (the bug tracker) for "easy" issues you may want to
  help solving, or even just reproducing:
  https://trac.ffmpeg.org/
  Also check there to see whether your favorite nit is already
  mentioned.
- Check the ffmpeg-devel archives, whether someone has already
  attempted what you are trying.
- Enjoy. :)

Cheers,
Moritz
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Re: [FFmpeg-user] frame rates

[Moritz Barsnick]

On Mon, Oct 04, 2021 at 09:15:49 +1100, Tim Lloyd wrote:
> I don't know whether I am expecting too much. I would like to lower the
> frame rate and hence the bit rate while keeping other video aspects the
> same. I decided to ramp down to 25 fps.

Depending on the content, that may not reduce the bitrate much at all.
But worth a try.

> The first attempt changed height &
> width so I added the "-vf" qualifier. This is me clutching at straws so time
> to seek assistance.
>
> ffmpeg -i file -r 25 -vf "scale=960:540" output
>
> This gives me:
>
> Duration : 58 min 59 s
> Bit rate : 854 kb/s
> Width    : 960 pixels
> Height   : 540 pixels
> Display aspect ratio : 16:9
> Frame rate mode  : Constant
> Frame rate   : 25.000 FPS
>
> I was expecting the bit rate to drop but that seems a huge drop. Is this to
> be expected? Is there a recommended path to lower the frame rate?

This is because ffmpeg encodes at its own defaults, in this case
probably (we would know if you showed us the complete console output)
to H.264 with CRF=23, preset medium. If the original was encoded for
more quality, it will have had that much more bitrate.

If the visual result of the conversion is okay for you, you can
consider yourself lucky, and enjoy the saved bandwidth/space. If you
would like more quality, explicitly set the crf to a lower(!) value,
and see what it changes.

Cheers,
Moritz
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Re: [FFmpeg-user] Latency in webcam capture

[Moritz Barsnick]

On Thu, Oct 07, 2021 at 16:41:04 +0200, Arif Driessen wrote:

> I've also tried:
>
> ffmpeg \
> -hide_banner \
> -analyzeduration 1 \
> -probesize 32 \
> -fflags -nobuffer \

I believe this explicitly disables the nobuffer option. Try
  -fflags +nobuffer

Have you tried using this flag directly with ffplay?

Moritz
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Re: [FFmpeg-user] How to extract audio from video without re-encoding?

[Moritz Barsnick]

On Mon, Oct 11, 2021 at 10:49:15 +0200, Android PowerUser wrote:
> Android PowerUser  schrieb am Mo., 11. Okt. 2021,
> 10:38:

> -vn -c:a copy

Agree.

[...]
> M4A is probably the best container for all expenses.

I consider M4A quite restricted:

https://en.wikipedia.org/wiki/Comparison_of_video_container_formats#Audio_coding_formats_support

Matroska (MKV) is much more flexible, as is ffmpeg's own NUT format.

Cheers,
Moritz
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Re: [FFmpeg-user] [EXTERNAL] Re: Parse a transport stream and get just 1 program

[Moritz Barsnick]

Hi Jim,

On Tue, Oct 12, 2021 at 12:27:15 +, FFmpeg user discussions wrote:
> I wonder what do you see that tells that the mapping is map 0:2 -map 0:4

This part:
> >   Program 1
> > Stream #0:2[0x51]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), 
> > yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 
> > 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
> > Side data:
> >   cpb: bitrate max/min/avg: 17598000/0/0 buffer size: 7995392 
> > vbv_delay: N/A
> > Stream #0:4[0x54](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
> > 5.1(side), fltp, 384 kb/s
> > Stream #0:10[0x56](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
> > stereo, fltp, 192 kb/s

It's the "#0:2", "#0:4", and so on, in the quoted text. You should also
be able to map by program ID:

$ ffmpeg ... -map 0:p:1 ...
or by the the PIDs - they are the "0x51", "0x54" and "0x56" seen above.

Cheers,
Moritz
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Re: [FFmpeg-user] unable to git pull, git clone due to SSL cert errors

[Moritz Barsnick]

Von: "Pavel Koshevoy" 
> Am I the only one having this problem:
> 
> $ git pull
> fatal: unable to access 'https://git.ffmpeg.org/ffmpeg.git/': SSL
> certificate problem: certificate has expired

Recent changes to the certificate chain of LetsEncrypt expose problems with 
gnutls. (git uses curl, which in turn uses gnutls on many platforms, though it 
can use openssl or mbedtls as well).

Check the dependencies and whether you have an update for gnutls on your 
platform. You may also need the latest ca-certificates package. On Ubuntu 
16.04, this is the relevant change to libgnutls:

gnutls28 (3.4.10-4ubuntu1.9) xenial; urgency=medium

  * Backport patches from Upstream/Debian to check validity against system
certs. This is to allow correctly validating default letsencrypt
chains that now also include a redundant expired certficate. LP:
#1928648

 -- Dimitri John Ledkov   Fri, 27 Aug 2021 
14:19:17 +0100

Not sure what the state on other distributions or OSs is.

Cheers,
Moritz
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Re: [FFmpeg-user] Quality Reduced when Burning Subtitles

[Moritz Barsnick]

On Sun, Sep 05, 2021 at 13:43:22 +, Veronica & Stephen McGuckin wrote:
> ffmpeg -i c:\video\input.mkv -filter_complex "[0:v][0:s]overlay[v]" -c:v 
> libx265 -pix_fmt yuv420p10le -profile:v high  -x265-params "-crf=10 -film 
> -hdr10+=1 -preset slow" -map "[v]" -map 0:a:0 -c:a copy c:\video\output.mkv

Please show us the complete, uncut console output of the command. It
would e.g. show which CRF actually takes effect.

I would actually expect an error from this option:
  -x265-params "-crf=10 -film -hdr10+=1 -preset slow"
That's not the correct syntax for x265-params.

https://ffmpeg.org/ffmpeg-codecs.html#Options-32
Quote:

Set x265 options using a list of key=value couples separated by ":". See 
x265 --help for a list of options.
For example to specify libx265 encoding options with ‘-x265-params’:
  ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

Perhaps you meant to use those "x265-params" as actual ffmpeg options.

> When the conversion is complete the output file is 143 mb with a bit rate of 
> 279 b/s.

I honestly doubt the 279 b/s. That would mean your video is 4299537
seconds long (if you mean 143 MB size - 537442 seconds if you mean 143
Mb). Did you mean 279 kbit/s?

Cheers,
Moritz
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Re: [FFmpeg-user] Can ffmpeg record video from this kind of URL?

[Moritz Barsnick]

On Wed, Sep 01, 2021 at 08:14:02 +0200, Bo Berglund wrote:
> How do I extract the proper m3u8 URL from the webpage in the best/safest way?
> Some of these pages display a scrolling long list of items when one presses 
> the
> F12 function key in FireFox and it is not obvious to me what I should try.
> And there are also URL:s ending with .ts in that scrolling list, but these
> change all the time so they are probably just short sections of the video,
> right?
>
> Is there a command line call I can make to extract the m3u8 URL automatically 
> so
> it can be used in a script only supplied the main page URL?

There's often a lot of magic involved in extracting the video URLs from
webpages - parsing JavaScript, downloading and parsing JSON files,
re-using cookies and referrers, and so on. Please do use the tool
youtube-dl, or the maintained fork yt-dlp, to extract these URL or even
to download the actual videos. The tool figures out the formats, the
URLs, and often even the de-encryption (not DRM though) of many many
web pages. (It can even call ffmpeg to download, and will do so by
default in many cases.)

To get the (or a) URL:
$ youtube-dl -g http://website-of-choice/page

If youtube-dl doesn't support a certain page, I load the page in Chrome
and look for the original file or m3u8 on the Network tab of the
developer console (Ctrl+Shift+C), and then pass that to ffmpeg.

Cheers,
Moritz
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Re: [FFmpeg-user] Next major release or point release for 4.4

[Moritz Barsnick]

Hi Bishal,

On Tue, Aug 24, 2021 at 03:42:50 +, DEBBARMA, Bishal via ffmpeg-user wrote:
> I am looking for  the next major / point release date for 4.4 that
> contain this patch @
> https://github.com/FFmpeg/FFmpeg/commit/26d3c81bc5ef2f8c3f09d45eaeacfb4b1139a777

The next major release is far away, yet this commit has been
back-ported to the 4.4 branch, so it will be in the coming 4.4.1
release. There is no date set yet for that release though.

Cheers,
Moritz
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Re: [FFmpeg-user] Why FFMPEG?

[Moritz Barsnick]

On Sun, Aug 15, 2021 at 21:28:12 +0200, Android PowerUser wrote:
> If FFMPEG is a collection of codecs then why are the most memory efficient
> codecs not available. The most memory-efficient waveform encoder xhe-aac,
> the most memory-efficient video codec VVC, the most memory-efficient image
> codec AVIF

What are you trying to say? Which media framework are you using instead
which supports these?

> ISOBMFF as in HEIF), not even the most memory-efficient vocoder Codec2 can
> be used without other downloads.

"without other downloads"? If you mean libocdec2: Some other team made
an effort to create a usable library, and ffmpeg makes use of it.
Where's the problem? The same is the case for H.264 via libx264 and
HEVC via libx265. (ffmpeg has no native encoders for these either.)

> So a cross-device codec collection is a good idea, but then you should do
> it right. I can't help with that, but those who can and have so far
> partially done it should end it.  Otherwise it doesn't make any sense ...
> right?

Creating and integrating codecs is not done easily. If you can't
contribute with code, you can surely sponsor a group of developers.

It does make sense though: HEVC is still being established, so
widespread use of VVC seems far away. Sure, a leading codec would be
nice, and I'm sure it's in the making, but it doesn't just appear
magically.

Cheers,
Moritz
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Re: [FFmpeg-user] 4K Way to large to stream

[Moritz Barsnick]

On Tue, Jun 29, 2021 at 08:57:51 -0400, rbar...@njdevils.net wrote:

> I would very much like to figure a way to Keep the 4K quality and reformat
> the video for streaming.  As an example, my daughters 2 hour recital hit the
> 80GB mark using a GoPro 9 set to record HDR.
> My ultimate goal is to have a 4K SDR video small enough to stream over PLEX.
>
> If I can avoid transcoding and keep it direct play (8 Bit, 5.1, SDR) then
> anything under 15GB will stream without buffering.

Unless that 80 GB file has 65 GB overhead (i.e. stuff not part of the
codec), you cannot achieve 15 GB without re-encoding. With re-encoding.
you cannot preserve the quality - reducing a video's size always
means reducing quality, unless you accept "reduce the quality so much
that I don't notice". And that latter subjective view is pretty much up
to you. Would you accept half the resolution (i.e. do you really need
4k on your screen)? Would you accept somewhat worse encoding quality?

It's pretty much up to you, how much reduction in quality you are
willing to accept. And to experiment whether e.g. libx264 with CRF 25
is sufficient, and brings any size reduction.

For example, I - for myself - can reencode my TV recordings (satellite
dumped to MPEG-TS on disk) from ~10 MB/s to 3 MB/s without noticing a
deterioration, in most cases. That would be acceptable to me.

Cheers,
Moritz
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Re: [FFmpeg-user] lot of open file handles

[Moritz Barsnick]

Hallo Wolfgang,

On Wed, Jun 23, 2021 at 06:08:30 +0200, Wolfgang Meyerle via ffmpeg-user wrote:
> Does anybody know why ffmpeg is using a lot of open file handles on the same
> filen when processing it?
>
> When I open up an mkv file and check with lsof the before and after count I
> get over 3000 file handles open on the same file.

I cannot reproduce this. What is the actual ffmpeg command you are
using, and how are you checking the open files?

> Is thtat normal or an ffmpeg bug?

3000 times on the same file isn't normal.

Moritz
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Moritz Barsnick]

On Mon, May 24, 2021 at 22:26:03 +0200, Moritz Barsnick wrote:
> MPEG-TS and demuxing isn't the problem. If you look at the ffmpeg
> output in the original e-mail from decoding the file, it correctly
> identifies the codec as "aac (LC)". The issue is the encoding to MP4.

On the other hand, when I copy AAC LC audio from MPEG-TS to MP4, it
works. So my analysis has some major flaw.

Moritz
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Moritz Barsnick]

On Wed, May 19, 2021 at 12:17:38 +0200, Tobias Rapp wrote:
> Looks like the data that should be found in the "esds" atom is missing or
> incomplete when using -c:a copy. See this link for details on how AAC
> profile information is stored in MP4:
>
> https://stackoverflow.com/questions/3987850/mp4-atom-how-to-discriminate-the-audio-codec-is-it-aac-or-mp3

Interesting stuff.

> Don't know enough about TS (de)muxing, though, to find out where this
> profile data should be parsed out from the input by FFmpeg.

MPEG-TS and demuxing isn't the problem. If you look at the ffmpeg
output in the original e-mail from decoding the file, it correctly
identifies the codec as "aac (LC)". The issue is the encoding to MP4.

The reason is that the AAC variants have only one codec tag
AV_CODEC_ID_AAC, and that needs to map to various tags, e.g. 0x40 for
AAC and 0x67 for AAC LC.

https://github.com/FFmpeg/FFmpeg/blob/3749eede66c3774799766b1f246afae8a6ffc9bb/libavformat/isom.c#L34

When writing the tag in the esds atom, there's no additional info
available allowing mapping to 0x67, therefore 0x40 is written:

https://github.com/FFmpeg/FFmpeg/blob/3749eede66c3774799766b1f246afae8a6ffc9bb/libavformat/movenc.c#L711

I have no idea how this info could be carried forward.

Moritz
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Re: [FFmpeg-user] Modify FFMP Vendor FourCC for ProRes Videostream

[Moritz Barsnick]

Hallo Marcel,

On Wed, May 19, 2021 at 11:48:58 +0200, Marcel Hinsch wrote:
> Looking into FFMPEG's source when trying to figure out how to change a
> ProRes Video Stream's vendor tag (which is required for certain automated
> spec compliency tests), I have found that the vendor code is hard-coded to
> "FFMP" as visible on this line of code:

Which spec compliancy demands that the file be written by Apple, Inc.?

> https://github.com/FFmpeg/FFmpeg/blob/c75dfa043585db7f5e20f5cab2ca8be1771ca7b6/libavformat/movenc.c#L2128
> . Being hardcoded, it doesn't seem like it can be changed unless the
> source is forked. Is there a reason this is set to "FFMP"? Would there be
> an issue changing it to "appl"? Why was a different value from "appl"
> chosen in the first place?

I guess it was chosen as such, because ffmpeg isn't Apple, and even
Apple specifies:

https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html

  Vendor
A 32-bit integer that specifies the developer of the compressor
that generated the compressed data. Often this field contains
'appl' to indicate Apple, Inc.

"Often" tells me a compliance tool shouldn't demand this value.

On the other hand, such tags can often be changed within ffmpeg. Just
not this one.

Cheers,
Moritz
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Re: [FFmpeg-user] Python / Qprocess misfires commands.

[Moritz Barsnick]

On Tue, May 18, 2021 at 12:05:40 +0200, Bouke wrote:
> > Then you're capturing your output incorrectly.

> Nah, this was an apology for an earlier message where I ment ’nothing except 
> the banner'

Hmm. Your attached reports indicate that there were more log messages.
I wonder where they went...

> > If QProcess terminates it unconditionally, there's nothing we can do(?).
>
> I understand, but now it’s the blame game, what happens where / when / why?

Sure. I don't understand QProcess (yet, or anymore), so I have no idea
what to debug. I don't know what to make of the return code 6 either.
it doesn't come from ffmpeg, as far as I can tell from the latter's
source.

ffmpeg isn't stuck at that point, it just terminates, right?

> > If ffmpeg crashes, that's bad. You would have to convince your system
> > to create a core dump, and would have to look at a backtrace. (Or run
> > ffmpeg within gdb.)
>
> Eeh, I’m a simple scripter, not a coder, no clue how to do that...

(If you can operate QProcess, you're no longer a "simple" scripter.)

Well, if MacOS happens to have gdb, it would work by replacing your
ffmpeg command line with:

$ gdb -ex r -ex bt --args ffmpeg [further command line options]

i.e. prepending gdb. The ffmpeg binary should not be stripped for this
to work properly.

MacOS may have lldb instead of gdb, but I have no experience in
operating that.

> > Do you need to run it in bash? Can you not set cmd to "ffmpeg" and
> > properly pass the arguments?
>
> Normally I do that, especially as that saves tons of headaches on escaping 
> stuff.
> But this was a suggestion from the QT forum, hoping it would provide a bit 
> more insight.
> https://forum.qt.io/topic/126674/escaping-colon-in-command-line-argument/8 
> 
>
> Trust me, it makes no difference if I use separate args or run it as bash.

I trust you. I was just speculating.

Moritz
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Re: [FFmpeg-user] Python / Qprocess misfires commands.

[Moritz Barsnick]

On Mon, May 17, 2021 at 09:19:46 +0200, Bouke wrote:
> > You will have to show/look at the console output of both commands.
> > (It's there for a reason, and most likely says what the issue is.)
>
> I know, but there was ’nothing’ on stdErr / stdOut.

Then you're capturing your output incorrectly.

> It turns out that FFmpeg crashes, nothing wrong with the command line.
> QProcess sees an error, and either QProcess terminates FFmpeg (I don’t think 
> so) or FFmpeg crashes by itself.

If QProcess terminates it unconditionally, there's nothing we can do(?).
If ffmpeg crashes, that's bad. You would have to convince your system
to create a core dump, and would have to look at a backtrace. (Or run
ffmpeg within gdb.)

> in Python:
> cmd = "bash"
> args = ["-c", "/Applications/ffmpeg -loglevel trace -report -f avfoundation 
> -i :0 /Volumes/Data/test.mov"]

Do you need to run it in bash? Can you not set cmd to "ffmpeg" and
properly pass the arguments?

Regards,
Moritz
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Moritz Barsnick]

On Mon, May 17, 2021 at 23:34:45 +0100, Simon Brown wrote:
> Thank you for your reply.  If it is the bitstream filter could you point me
> to the relevant source file that is responsible for this filter?

Sure, it's here:
https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/aac_adtstoasc_bsf.c

> mp4info doesn't deal with TS, as you surmise.  FFMpeg reports the output
> file as AAC (not LC).  I'm not sure VLC gives that detail.  But given that
> the ffmpeg processor and the mp4info program both seem to concur when there
> is or isn't AAC-LC I'm not sure there is much value in trying a third
> tool.

You could also try mediainfo. I also found a tool capable of parsing
and deconstruction AAC headers (https://p23.nl/projects/aac-header/),
but I don't trust it since it isn't capable of recognizing e.g.
HE-AACv2 (though I haven't taken the time to understand all the headers
myself).

I couldn't reproduce the loss of this information with the input
samples I have. Can you share your input file? Since it's an MPEG-TS.
perhaps the first MB (or five) suffice.

> I will download the latest prebuilt binaries for Windows and give it a try
> with those.  I'd still be interested in the source file responsible as the
> version on the embedded system I'm using is not the most recent build, and
> rebuilding for that isn't something I want to do at this stage.

I understand.

Cheers,
Moritz
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Moritz Barsnick]

On Fri, May 14, 2021 at 14:31:42 +0100, Simon Brown wrote:
> Hi,
> I have a mpeg2 transport stream with video as H264 and audio as AAC LC.  If
> I use the following command to convert it to fragmented MP4 by just copying
> the encoded data, then the result is now AAC, and not AAC LC.  If instead I
> re-encode with AAC asking for profile:a aac_low then I get AAC LC.  But if
> the input source is AAC LC why would it change the output type to AAC?
>
> ffmpeg.exe -f mpegts -fflags +nobuffer+nofillin -probesize 500 -i
> soc_udp_rx_02.ts -c:a copy -bsf:a aac_adtstoasc -c:v copy -f mp4
> -frag_duration 8 -movflags +empty_moov+default_base_moof -metadata
> title="media source exentions" testaudio.mp4

This shouldn't change anything in the AAC stream, unless the bitstream
filter is capable of ruining it.

> ffmpeg version git-2020-06-19-2f59946 Copyright (c) 2000-2020 the FFmpeg 
> developers

This is a bit old, please retry with a recent build.

> For although there it says aac (LC) if I run mp4info on the resultant file
> I get this:
>
> mp4info testaudio.mp4
[...]
> Track 2:
>   flags:3 ENABLED IN-MOVIE
>   id:   2
>   type: Audio
>   duration: 0 ms
>   language: und
>   media:
> sample count: 0
> timescale:48000
> duration: 0 (media timescale units)
> duration: 0 (ms)
> bitrate (computed): 125.373 Kbps
> sample count with fragments: 848
> duration with fragments: 868354
> duration with fragments: 18091 (ms)
>   Sample Description 0
> Coding:  mp4a (MPEG-4 Audio)
> Stream Type: Audio
> Object Type: MPEG-4 Audio
> Max Bitrate: 128250
> Avg Bitrate: 0
> Buffer Size: 0
> Codecs String: mp4a.40.0
> MPEG-4 Audio Object Type: 0 (UNKNOWN)
> Sample Rate: 48000
> Sample Size: 16
> Channels:2

What does it say about the original? (Nothing, presumably, because it's
MPEG-TS.) What does ffmpeg say about the output file? Do you have any
other tool which can use to check?

Cheers,
Moritz
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Re: [FFmpeg-user] Python / Qprocess misfires commands.

[Moritz Barsnick]

On Fri, May 14, 2021 at 13:16:41 +0200, Bouke wrote:
> Using Python to fire FFmpeg
> cmd = ‘ffmpeg’
> args = [‘-f’, ‘avfoundation’, ‘-i’, ‘:0’]
>
> with a QProcess, this returns exit code 6, exitstatus 1
>
>
> ffmpeg -f avfoundation -i :0 in a Terminal works.
> (At least one output file….)

You will have to show/look at the console output of both commands.
(It's there for a reason, and most likely says what the issue is.)

Moritz
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Re: [FFmpeg-user] Failed / strange build script, now with log / script files in link

[Moritz Barsnick]

On Tue, May 11, 2021 at 09:50:14 +0200, Bouke wrote:
> Logfiles and script is here:
> Download link
> https://we.tl/t-UB5Ynvv8DE 
>
> (Original script comes from 
> https://github.com/markus-perl/ffmpeg-build-script 
>  )

You can't report your issue there, because you modified the script,
correct? I don't see a drawtext/fontconfig feature request there
either. (I do see your Decklink one.)

Anyway, regarding fontconfig:

> test_pkg_config libfontconfig fontconfig fontconfig/fontconfig.h FcInit
> pkg-config --exists --print-errors fontconfig
> Package fontconfig was not found in the pkg-config search path.
> Perhaps you should add the directory containing `fontconfig.pc'
> to the PKG_CONFIG_PATH environment variable
> No package 'fontconfig' found
> ERROR: fontconfig not found using pkg-config

Does your machine have actually fontconfig.pc?
On Linux, I would do
$ locate fontconfig.pc
I assume MacOS has a similar function. (Note that the database for
locate is updated in large intervals, it won't find anything I
installed today.)

And regarding static/dynamic:
> MacBook-Pro:~ bouke$ /Users/bouke/Desktop/ffplay ; exit;
> dyld: Library not loaded: /usr/local/opt/fontconfig/lib/libfontconfig.1.dylib
>   Referenced from: /Users/bouke/Desktop/ffplay
>   Reason: image not found

I have no idea why the compiler would link against a dynamic library
which isn't there. On Linux, I would know how to check the compile-time
and run-time linkers, but not here remotely.

And regarding static vs. dynamic: Is a static libfontconfig available
at all? Does the system provide static libraries? (I sort of doubt it.)

Cheers,
Moritz
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Re: [FFmpeg-user] Sometimes, ffplay fails showing raw video received over LAN

[Moritz Barsnick]

Hi Thomas,

On Tue, May 11, 2021 at 14:40:40 +0200, FFmpeg user discussions wrote:
> I grab video on a Raspberry Pi, host 1,  and sends raw video to another
> host, host 2, on the same LAN:
>
> Host 1:
>
> export FPS=5; export W=352; export H=240; \
>
> raspividyuv -t 0 -n -v -hf --exposure sports -fps ${FPS} -w ${W} -h ${H} -p
> 800,50,${W},${H} -o - | \
>
> ffmpeg -f rawvideo -video_size ${W}x${H} -r ${FPS} -i pipe:0 -f rawvideo
> udp://192.168.1.29:7000
[...]
> Every now and then video looks bad. I.e.. color is bad or there is unwanted
> pattern on the video.

Have you tried wrapping the video in nut or matroska containers?

Moritz
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Re: [FFmpeg-user] gnutls compilation error

[Moritz Barsnick]

Hi Uros,

On Wed, May 05, 2021 at 19:29:43 -0400, Uros Kolar wrote:
> pkg-config can find gnutls if I run it manually:
>
> $ pkg-config gnutls --cflags
> -I/usr/local/include
>
> I also exported GNUTLS_CFLAGS and GNUTLS_LIBS. I am not sure if the
> undefined references  are related with this error or not.

It appears to be using the static gnutls, so this fails:

> /usr/local/lib/libgnutls.a(kx.o): In function `_gnutls_nss_keylog_write':
> /home/ec2-user/ffmpeg-libfdk_aac/src/gnutls-3.6.15/lib/kx.c:153: undefined 
> reference to `pthread_mutex_lock'
> /home/ec2-user/ffmpeg-libfdk_aac/src/gnutls-3.6.15/lib/kx.c:163: undefined 
> reference to `pthread_mutex_unlock'

because the linker should be using the output of
$ pkg-config --libs --static gnutls
not
$ pkg-config --libs gnutls

Did you read these very last lines of your config.log?:

> ERROR: gnutls not found using pkg-config
> Note: When building a static binary, add --pkg-config-flags="--static".

That's what you should be doing.

Cheers,
Moritz
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Re: [FFmpeg-user] Failed / strange build script

[Moritz Barsnick]

On Mon, May 10, 2021 at 10:47:24 +0200, Bouke wrote:
> Hi all,
> I’ve downloaded a FFMpeg build script and altered it to my liking
> (Added Decklink support / drawtext)
>
> running the script as ./build-ffmpeg -b --enable-gpl-and-non-free
>
> Now it does build on a Catalina system, but fails on a Big Sur.
> The succesfull build however has dependencies  where everything should be 
> static.
>
> MacBook-Pro:~ bouke$ /Users/bouke/Desktop/ffplay ; exit;
> dyld: Library not loaded: /usr/local/opt/fontconfig/lib/libfontconfig.1.dylib
>   Referenced from: /Users/bouke/Desktop/ffplay
>   Reason: image not found

It doesn't seem like a good build script to me. Where did you get it
from? Why not use a well-established ffmpeg configure command line?

> What am I missing?
> Logfiles and script are attached

No, they are not.

Cheers,
Moritz
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Re: [FFmpeg-user] convert a video to some format and than bring it back exactly the same

[Moritz Barsnick]

Hi Julien,

On Thu, Apr 29, 2021 at 04:17:20 +0200, Julien Dotsev wrote:
> I see there is a difference in the file size. The first avi is *1462KB* and
> after the reconversion is *1474KB*.
>
> My question is is there a way to take a video file from a client, convert
> it to some readable by  HTML  tag format and after I finish using it
> be able to bring to the client the original file, exactly the same.

No, there is no such way. Every video operation with ffmpeg involves
demuxing and muxing. And every muxer - so also ffmpeg's - has more or
less subtle differences. ffmpeg may be less efficient (therefore the
larger size), have other presents, and so on. Furthermore, your first
operation may also lose metadata or data streams, which your second
operation cannot recover.

Cheers,
Moritz
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Re: [FFmpeg-user] Can FFmpeg Record Video On Linux Using V4L2 in V4L2_CAP_VIDEO_CAPTURE_MPLANE Mode?

[Moritz Barsnick]

Hi Andrew,

On Wed, Apr 28, 2021 at 13:17:25 +, FFmpeg user discussions wrote:
> $ ffmpeg -f v4l2 -list_formats all -i /dev/video1  # Look at device 
> information
>
> $ ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video0 output.mkv  
> # save the video

You'll have to use the same device for both commands, but I assume
that's a typo or a copy/paste mistake.

> I get the following output with both commands
>
> [video4linux2,v4l2 @ 0xcbbd4f0] Not a video capture device.
> /dev/video1: No such device

You should show us the complete, uncut console output of the command
using the correct device.

Can you also use a tool such as v4l-info, and post its output here?

> I see here 
> (https://github.com/FFmpeg/FFmpeg/blob/master/libavdevice/v4l2.c#L172) that 
> v4l2 devices need to support V4L2_CAP_VIDEO_CAPTURE to open. My device does 
> not support V4L2_CAP_VIDEO_CAPTURE but it does support 
> V4L2_CAP_VIDEO_CAPTURE_MPLANE. Does FFmpeg have support for 
> V4L2_CAP_VIDEO_CAPTURE_MPLANE?

I'm not really too knowledgable about these "newer" devices, but I
believe they are supported by the v4l2_m2m codec(s). Someone else may
need to explain why your device isn't recognized, after you have
provided the requested output.

Here is a check for MPLANE:
https://github.com/FFmpeg/FFmpeg/blob/8bc7f69cefbcbb265b41717608dc8436dc864a55/libavcodec/v4l2_m2m.c#L51

Cheers,
Moritz
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Re: [FFmpeg-user] Can ffmpeg be set to wait if a stream stop happens?

[Moritz Barsnick]

Hej Bo,

On Wed, Apr 21, 2021 at 16:59:10 +0200, Bo Berglund wrote:
> But what I have noticed lately is that sometimes the stream it downloads from
> stops feeding data for a few seconds (don't know for how long) and it seems 
> like
> ffmpeg at that times finishes up the output file and exits.

ffmpeg shouldn't interrupt if the input stream just "stalls". It will
interrupt if the stream server actually closes the connection. With
HLS, where the URLs to open continuously change, ffmpeg will retry a
certain number of times (IIRC) before giving up.

> Is there a way to tell ffmpeg to stay online for that full amount even if the
> stream pauses for some reason?

I suspect it isn't "pausing". It would be practical to see a complete
log output, of course. (You can generate them by using the "-report"
option, but they may become large, unless you reduce the default report
log level.)

Cheers,
Moritz
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Re: [FFmpeg-user] Converting wma to mp3, avoiding joint stereo

[Moritz Barsnick]

On Sat, Apr 17, 2021 at 13:42:02 -0400, Alan Corey wrote:
> can deal with.  So I did the basic
> ffmpeg -i somefile.wma somefile.mp3
> And it works, but mpg123 tells me the file is joint stereo. It's
> quite possible the wma files are joint stereo, I don't know anything
> about them, but when I convert a flac file (to wav then mp3) I can at
> least nominally output true stereo.

ffmpeg's encoder is independent of the input, joint stereo is its
default.

> accomplish anything but I call LAME with -m s.  Is that possible with
> ffmpeg?  How?  Do I need to get into channel mapping?

As Paul wrote:
  -joint_stereo 0
(as output option).

> Joint stereo I remember from the early days of fm stereo that had to
> also be playable on mono radios.  Better than mono but not as good as
> true stereo.

It depends on the quality of the encoder, but basically, joint stereo
means that the encoder makes use of the similarity of the two channels
to optimize bandwidth use. At the same (fixed) bitrate, you may encouter
better quality in critical sections. With modern encoders (such as
mp3lame), you shouldn't notice a difference in your stereo separation.

In other words: Just continue to use joint stereo if you can.

See also
https://wiki.hydrogenaud.io/index.php?title=Joint_stereo#Additional_information

Cheers,
Moritz
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Re: [FFmpeg-user] Tokenised http subtitle source

[Moritz Barsnick]

Hi Adam,

On Wed, Apr 14, 2021 at 14:15:48 +0100, FFmpeg user discussions wrote:
> [https @ 0x7f40de879b40] HTTP error 403 
> Forbiddenhttps://some_domain.com/subtitles/18217/bf4130a3-64f8-40cb-aae8-720da75797a7.srt?AWSAccessKeyId=A*Z:
>  Server returned 403 Forbidden (access denied)

You should show us the complete, uncut console output, without the
access key of course.

> It looks like it is cutting off the URL after the aws access key.

It looks like you may be using Unix/Linux and your shell is
interpreting '&'. You may need to quote the complete URL on the command
line. But I am just guessing.

> Does anybody know if FFmpeg will work with this type of token or should I 
> just make a HTTP.get call in my script to localise the SRT file?

This should just work. ffmpeg's URL handling is limited to a length of
4096.

Cheers,
Moritz
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Re: [FFmpeg-user] How to convert video.ts to video.mp4 efficiently?

[Moritz Barsnick]

On Tue, Apr 13, 2021 at 17:49:55 +0200, Bo Berglund wrote:
> Question:
> Is there a quicker way to just go from ts to mp4 so the file format is OK even
> though the video size remains at 1280x720?
>
> (I could live with the size even though the disk storage takes a hit...)

Yes, by remuxing, i.e. using the "copy" codec for video as well:

$ ffmpeg -i %INFILE% -c copy %OUTFILE%

or, as Carl suggested:

$ ffmpeg -i %INFILE% -c copy -movflags +faststart %OUTFILE%

As mentioned, the suffix .mp4 already implies the format you are trying
to achieve.

Cheers,
Moritz
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Re: [FFmpeg-user] ERROR: libglslang not found

[Moritz Barsnick]

On Thu, Apr 08, 2021 at 09:49:02 +1000, Dylan Fernando wrote:
> g++ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE 
> -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC -std=c11 
> -fomit-frame-pointer -fPIC -pthread -D__STDC_CONSTANT_MACROS -c -o 
> /tmp/ffconf.j3rfnDQR/test.o /tmp/ffconf.j3rfnDQR/test.cpp
> ./configure: 984: g++: not found
> ERROR: libglslang not found

libglslang support apparently requires a C++ compiler, which you don't
seem to have.

Moritz
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Re: [FFmpeg-user] -b:v option

[Moritz Barsnick]

On Thu, Apr 08, 2021 at 11:53:52 +, Mar Andrés López wrote:
> I want to do this in order to generate a DASH content, one of the
> profiles sould have the original bitrate, and the other ones fixed
> values. As an example:

Have you considered using the copy codec for the first profile? That
will guarantee the same bitrate, and not induce a loss in quality.

Cheers,
Moritz
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Re: [FFmpeg-user] 'mix=nb_inputs=2' provokes error...

[Moritz Barsnick]

On Wed, Mar 24, 2021 at 10:18:55 +0100, Peter White wrote:
> > [AVFilterGraph @ 025772b53dc0] Error initializing filter 'mix' with 
> > args 'nb_inputs=2:weights=3 2'
> > Error initializing complex filters.
> > Option not found
>
> There seems to be an error in the documentation. The option is called
> "inputs".

Indeed.

BTW, to check options, you can view the auto-generated documentation:

$ ffmpeg -h filter=mix

Cheers,
Moritz
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Re: [FFmpeg-user] sytraming AAC+

[Moritz Barsnick]

Hi Robert,

On Mon, Mar 22, 2021 at 18:08:23 +1300, Robert Jeffares wrote:
> I am getting...
>
> [rtp @ 0x55e52ffe8c00] AAC with no global headers is currently not
> supported.

You are usually asked to provide the full, uncut console output of your
command.

> ffmpeg -i http://localhost:8000/wireless.aacp -acodec copy  -vn -flags
> +global_header -f rtp rtp://192.168.1.100:7001
>
> is there support for aac coming?

AAC is supported. Your input seems to be raw AAC, which ffmpeg cannot
(yet) add information about to RTP. Since you are copying, not
reencoding, the option "-flags +global_header", which is encoding
flags, has no effect.

You might try a different RTP format as suggested here:
https://video.stackexchange.com/a/24454
i.e. using
-rtpflags latm
but I don't know whether your RTP target supports this.

Cheers,
Moritz
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Re: [FFmpeg-user] Error decoding RTSP stream using h264_cuvid vcodec

[Moritz Barsnick]

On Wed, Mar 10, 2021 at 15:54:28 -0500, FFmpeg user discussions wrote:
> Oh duh! Thanks @Moritz!

You're welcome.

> As a side note, I just wanted to mention that we mostly deal with RTMP
> streams and we have been starting the ffmpeg process the way I described
> for over a year and half now and it never complained.

Hmm, I don't think an "h264_cuvid" encoder has ever existed - only a
decoder. (There was some cleanup recently in those aliases, but I
cannot identify such an encoder having being removed.) Perhaps you are
confusing the two?

> would be interesting to note the behaviour here. Thanks again!

No problem, as long as your encoding works again now.

Cheers,
Moritz
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Re: [FFmpeg-user] RTSP to Twitch

[Moritz Barsnick]

Hi Sébastien,

On Wed, Mar 10, 2021 at 12:55:25 +0100, Sébastien Bonaimé wrote:

> included x264 codec are : libx264 libx264rgb h264_omx h264_v4l2m2m h264_vaapi

> ffmpeg  -probesize  10M  -rtsp_transport tcp -thread_queue_size 1000k 
> \
>   -i $url \
> -vsync 0 -enc_time_base -1 \
>  -bufsize 1024k -preset "$QUAL" \
> -b:v $CBR  -minrate $CBR -maxrate $CBR   -keyint_min 24 -g 24 
> \
> -crf 10  -c:v libx264\
> -f flv $twitch_url
>
> With camera in CBR mode, I can't reach 25 FPS but only 14. And the
> stream to twitch does not last more than ew seconds. What can I do to
> improve the command ?

- Do you really need to reencode for twitch? Does it need CBR?
  Otherwise, you could just use the "copy" codec.

- AFAIU, you cannot combine "-b:v" and "-crf". I believe (fro mlooking
  at your logs) that the latter is being ignored.

- CRF 10 is really really low. What are you trying to achieve? Is the
  default of CRF 27 not good enough? CRF 10 is bound to take a lot of
  additional compute power (if not ignored).

- You listed other available encoders. Have you tried any of them? (I
  have no experience with HW accel on RaspPi, others may pitch in.)

> FFmpeg version is
> ffmpeg version 4.1.6-1~deb10u1+rpt1 Copyright (c) 2000-2020 the FFmpeg 
> developers

You should try to get hold of a newer version of ffmpeg. You may have
to compile yourself though. A newer version probably won't help with
the speed of libx264, but the support for the other HWAccels may have
improved.

Cheers,
Moritz
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Re: [FFmpeg-user] Error decoding RTSP stream using h264_cuvid vcodec

[Moritz Barsnick]

On Tue, Mar 09, 2021 at 16:46:41 -0500, Nishant Trivedi via ffmpeg-user wrote:
> I am trying to decode a RTSP video stream using ffmpeg while using
> h264_cuvid vcodec to take advantage of hardware acceleration on a machine
> with NVIDIA GPU. I am seeing an error with the message Unknown encoder
> 'h264_cuvid'.

That's because no such encoder exists.

Have you tried "h264_nvenc"?

> ffmpeg version n4.1.6-5-g7f0db52c53 Copyright (c) 2000-2020 the FFmpeg 
> developers

On this list, it's also always a good idea to use a recent version of
ffmpeg - preferred from latest git. That way, you can rule reporting
bugs or missing features which are no longer valid.

Cheers,
Moritz
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Re: [FFmpeg-user] How to get he total number of frames

[Moritz Barsnick]

On Tue, Mar 09, 2021 at 11:07:21 +0100, Ulf Zibis wrote:
> Am 09.03.21 um 10:43 schrieb Ulf Zibis:
> > I suspect that the last "progression" line shows the exact count,
> > as it may only report the last progression state a few µs before
> > the operation is completed.
>
> Can one please clarify this?

The progess line is printed periodically, and additionally after the
end of processing. Therefore the last visible line does not omit any
stats.

Moritz
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Re: [FFmpeg-user] repeat a frame

[Moritz Barsnick]

On Thu, Mar 04, 2021 at 03:57:38 -0500, Mark Filipak (ffmpeg) wrote:
> Well, per rational.h, 'num' & 'den' are both integers.
>
> Now, I don't know how '72' is stored. Is it stored as an int64?

They are defined as "int", which can be platform dependant. ffmpeg
assumes/guarantees ints to be at least 32 bits.

> I don't know, but I do know that it can't be stored as an 8-bit
> integer or a 16-bit integer or anything else that has 8-bit or 16-bit
> resolution.

If you mean the number 72, that's correct.

> That includes AVRational. 72 can't be stored as
> AVRational. AVRational just doesn't have enough resolution.

Huh? You didn't even wait for the answer to how it is stored. And you
even asked whether it was 64 bits. This would work just fine with 64
bits.

It's a rational number of two integers with at least 32 bits. 1/72
is probably stored as { 1, 72 }, but could also be { 2, 140 }
for example, unless they get reduced automatically (I'm too lazy to
check).

> I'm sure you know that the running time produced by 'PTS's depends on frame
> rate *and* time_base. For example, for 23.976fps and TB = 1/(72 ticks/s)
> = 1.3[8..] µs/tick, a PTS of +9223372036854775800 indicates a
> 307138595965860 maximum frame number.

Huh? At 72 ticks/seconds, and PTS being number of ticks, you can
create timestamps spanning
(2^63 - 1) / 72 seconds
which is
~12810238940076 seconds
which is
~40 years.

> Do you agree with my figures? Do you see the difference between time
> extents and time resolution?

I don't agree with your figures.

Moritz
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Re: [FFmpeg-user] .m3u8 video being downloaded converted to mp4 result in trouble

[Moritz Barsnick]

On Wed, Mar 03, 2021 at 22:35:56 +, Budi wrote:
> Why did the .m3u8 video being downloaded converted outright to mp4
> result in lower resolution quality?
> It's anyway either done by vlc or ffmpeg cli

Pease show us te ffmpeg command you are using and its complete, uncust
console output.

Thanks,
Moritz
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Re: [FFmpeg-user] split files based on time

[Moritz Barsnick]

Hi Raj,

On Tue, Mar 02, 2021 at 21:14:23 -0700, Raj Cryp wrote:
> ffmpeg has the option to split file based on time. Say starting with time t
> and go for length l.
>
> I am thinking of basically computing the time windows which will result in
> the desired file size. Say a 1 min video of 20 MB will need to be split in
> parts roughly 6 sec each.
>
> Is there an option to pass desired file size directly into the command?

You should take a look at the segment muxer and its examples:
https://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment

It can split at regular or arbitrary time points.

Cheers,
Moritz
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Re: [FFmpeg-user] repeat a frame

[Moritz Barsnick]

On Tue, Mar 02, 2021 at 17:32:42 -0500, Mark Filipak (ffmpeg) wrote:
> Thank you, Jim. To the best of my knowledge, rational is not a 'C' datatype.

No, but it is an ffmpeg data type, AVRational, as defined in
libavutil/rational.h, along with functions to operate on this type.

> I need to know the dimension (or 'C' datatype) of TB as it is used in PTC
> calculations.

Do you really need to know?

Cheers,
Moritz
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Re: [FFmpeg-user] Issue when adding a title to an audio stream within a mp4 file

[Moritz Barsnick]

On Mon, Mar 01, 2021 at 15:17:40 +0100, PPRJ01 wrote:
> But ... I don't understand what you are planning to do.

I'm not planning to do anything. I checked whether this could easily be
fixed, but it cannot with my knowledge.

> Please tell me if I'm right or wrong :
> - ffprobe/ffmpeg will display the data contained in the "name" atom to 
> stdout/stderr

No. When reading a file, ffmpeg does not understand it.

> BUT
> - ffmpeg will NOT be able to copy the audio stream "title" to a new MP4 
> output file by using the -c copy option

Correct, ffmpeg does not understand your title when reading, and can
therefore also not convey it to an output file.

An ffmpeg developer, especially one who knows their way around atoms,
might be able to fix this. (I can give more details what my issues
were, if anyone picks this up.)

Moritz
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Re: [FFmpeg-user] aax to m4b with chapters

[Moritz Barsnick]

On Fri, Feb 26, 2021 at 12:30:10 +, Max Blindenhoefer wrote:

> I read the Documentation, but didn’t understand everything. What am i
> supposed to insert after the command „-map_chapters“ and what is
> meant with „Input_file_index“?

Th input file index is explained e.g. in this chapter:
https://ffmpeg.org/ffmpeg.html#toc-Advanced-options

Basically, for your first (or only) input, use "0" (but that's the
default, apparently). Or higher numbers for your further inputs.

If you only have one input, and chapters aren't being mapped, then
something else is not working.

Moritz
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Re: [FFmpeg-user] Issue when adding a title to an audio stream within a mp4 file

[Moritz Barsnick]

On Wed, Feb 24, 2021 at 17:37:18 +0100, PPRJ01 wrote:
> I observed that the command below seem to work properly when adding a title 
> to an audio stream within an MP4 file :
>
>   ffmpeg -i INPUT.mp4 -c copy -metadata:s:a:0 "title=mono" OUTPUT.mp4
>
> Both VLC and Exiftool do display correctly the audio stream title of the 
> OUTPUT.mp4 file when needed.
>
> The point is that neither ffmpeg nor ffprobe can do it.

Indeed, I can confirm this. ffmpeg is able to insert the per-track
"title", but not able to extract it.

(Technical detail: it's the "name" atom beneath the "udta" atom beneath
the "trck" atom.)

> Maybe this is a designed behavior of ffmpeg/ffprobe.
> Or a bug ?

A shortcoming.

I'm able to change the ffmpeg source code to parse the per-track title,
but cannot get it inserted properly in the demuxer metadata.

Moritz
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Re: [FFmpeg-user] Do I really need libaom?

[Moritz Barsnick]

On Tue, Feb 23, 2021 at 10:59:07 +0300, Alexander Dyagilev wrote:
> But do I really need to encode?

That depends on what you want to achieve. If you use the copy codec
("-c copy"), you get a much faster creation of the target file, but
with unchanged codecs. You write:

> to convert arbitrary video files to MP4

MP4 is just a container format. If the consumers of your files can
decode AV1, then fine.

You would need to re-encode if the source file contains codecs not
supported by MP4.

Cheers,
Moritz
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Re: [FFmpeg-user] Re-writing hot .ts file that rotates every 24 hours in realtime

[Moritz Barsnick]

On Wed, Feb 17, 2021 at 22:18:00 -0800, Carl Zwanzig wrote:
> On 2/15/2021 11:33 PM, Korn Moffle wrote:
> > What I am trying to do:
> > a. copy the file in realtime ? So, ffmpeg would be rewriting file1.ts into 
> > file2.ts as file1.ts is constantly being written into.
>
> What's the goal of that?

I think he may be trying to tell us he wants to process the data
created by the first ffmpeg in realtime?
>
> > b. tell ffmpeg to read the file with a constant 3 second delay from the 
> > most updated frame?
>
> ...to add a 3 second delay?

Perhaps to avoid reading beyond the end of the file?

It sound to me as the first process is constantly recording a stream
straight to disk, and shall turn around to the next file after a
certain time. The second process reads these files while they are being
created, and does something else.

This sound a lot live HLS live streaming. The first process creates a
dynamic .m3u8 with segments, the second process knows how to read
these in real time. It even uses MPEG-TS underneath by default.

Have a look at the "hls" muxer (and demuxer).

Cheers,
Moritz
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Re: [FFmpeg-user] compiling ffmpeg for nvidia gpu under Debian Bullseye

[Moritz Barsnick]

On Wed, Feb 10, 2021 at 13:53:32 -0500, Steve Newcomb wrote:
> about what happened and how I eventually succeeded.  (I did everything as
> superuser, so you won't see any "sudo" below.)

I do see sudo below. And I don't see any reason to do any steps beyond
installation as superuser or with sudo. Don't compile as root!

> cd nv-codec-headers && sudo make install && cd –
> This worked, but the final invocation of cd was uninterpretable due to the
> last character's being garbled (by my browser? or by Emacs?).  I followed
> this incantation with a "cd ..", which turned out to be the right thing to
> do.

In nvidia's instruction, the web code apparently inserted some fancy
dash '–' instead of the regular '-'.

> apt-get install build-essential yasm cmake libtool libc6 libc6-dev unzip wget 
> libnuma1 libnuma-dev
> This turned out not to be enough packages.  I ended up installing the 
> following packages, as well, even if some of them may not be strictly 
> necessary.
> I always install lots of utility packages which don't appear on this list, so 
> this list, too, may not be complete:
> nvidia-detect
> nvidia-driver
> libnvidia-encode1
> libnppc11
> libnppial11
> libnppicc11
> libnppidei11
> libnppif11
> libnppig11
> libnppim11
> libnppist11
> libnppisu11
> libnppitc11
> libnpps11
> nvidia-cg-toolkit
> nvidia-cuda-gdb
> nvidia-cuda-toolkit
> nvidia-cuda-toolkit-gcc
> nvtop

These all look like nvidia's dependencies. Could it be that they are
part of the mentioned "CUDA toolkit", as listed under "Prerequisites"?
(Though if you can get them from the system repos, I think's that's
fine. Just perhaps not what nvidia wants you to do.)

> ./configure --enable-nonfree -–enable-cuda-sdk –enable-libnpp
> --extra-cflags=-I/usr/local/cuda/include
> --extra-ldflags=-L/usr/local/cuda/lib64
> This incantation had 2 problems.
>
> (1) --enable-cuda-sdk is an obsolete argument.  I changed it to 
> --enable-cuda-nvcc
>
> (2) The configure failed with the report: "ERROR: failed checking for nvcc."  
> Why?
>     The configure script checks nvcc by passing itan obsolete architecture
> code, "compute_30", which modern nvcc does not support.  
> 
> I edited the part of the configure script that says:
> if enabled cuda_nvcc; then
> nvcc_default="nvcc"
> nvccflags_default="-gencode arch=compute_30,code=sm_30 -O2"
> else
> nvcc_default="clang"
> nvccflags_default="--cuda-gpu-arch=sm_30 -O2"
> NVCC_C=""
> fi
> ...so it instead said:
> if enabled cuda_nvcc; then
> nvcc_default="nvcc"
> nvccflags_default="-gencode arch=compute_75,code=sm_75 -O2"
> else
> nvcc_default="clang"
> nvccflags_default="--cuda-gpu-arch=sm_75 -O2"
> NVCC_C=""
> fi
> ...although apparently it would have been fine to change "30" to "50" 
> instead of "75" as I did.
> Perhaps ffmpeg developers should make this change, or do something 
> similar.

Indeed, perhaps ffmpeg developers need to look at this.

Did you give feedback to nvidia regarding their document?

Thanks,
cheers,
Moritz
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Re: [FFmpeg-user] Stack + resize + use audio from first video?

[Moritz Barsnick]

On Mon, Feb 08, 2021 at 13:00:15 -0600, Gilles wrote:
> In one command, I'd like to stack two videos, resize them to the same
> width+height, and use audio from the first video.

Which of the below works for you? The first two, right?

> #Stack
> ffmpeg -i input0 -i input1 -filter_complex vstack=inputs=2 output
>
> #Using audio from one particular input. This example will use the audio from
> input1:
> ffmpeg -i input0 -i input1 -filter_complex "[0:v][1:v]vstack=inputs=2[v]"
> -map "[v]" -map 1:a output
>
> #Resize/scale an input
> ffmpeg -i input0 -i input2 -filter_complex
> "[0:v]scale=640:-1[v0];[v0][1:v]vstack=inputs=2" output

You could just merge your work from the second and the third example.
Did you try this?:

$ ffmpeg -i input0 -i input1 -filter_complex 
"[0:v]scale=640:-1[v0];[v0][1:v]vstack=inputs=2[v]" -map "[v]" -map 1:a output
(Untested by myself.)

There's also a very handy filter called "scale2ref". It will scale one
input based on the other's size:

https://ffmpeg.org/ffmpeg-filters.html#scale2ref

Cheers,
Moritz
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