Re: [FFmpeg-user] a couple of things to look at

2024-03-02 Thread Greg Oliver
On Sat, Mar 2, 2024 at 9:52 PM Mark Filipak 
wrote:

> I have a couple of things to look at.
>
> https://markfilipak.github.io/Video-Object-Notation/Streams.html
>
> https://markfilipak.github.io/Video-Object-Notation/GOP%20%26%20Frame%20Reordering.html
>
> Comments are welcome. Please be brutal. 'Streams' is crucial.
>
> I have little knowledge of transport streams beyond being 188 bytes/packet
> and beginning with 0x47.
> For example, if there are transport streams, are there program streams
> inside them or not? I would
> guess there are but I don't have a single TS to evaluate.
>

I have tons of US Broadcast TS streams if you need one, ping me and I'll
share a link to one.  These are directly from a Silicon Dust HDHR, so
should be unaltered.  I can send you a PBS stream since they have no
problem with re-transmission.



>
> Thanks --Mark.
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Re: [FFmpeg-user] Packet corrupt on cut, notice on concat

2024-02-18 Thread Greg Oliver
On Sun, Feb 18, 2024, 11:04 AM Mark Filipak 
wrote:

> On 18/02/2024 11.53, Gyan Doshi via ffmpeg-user wrote:
> > On 2024-02-18 09:59 pm, Mark Filipak wrote:
> >> On 18/02/2024 05.09, Gyan Doshi wrote:
> >>>
> >>> Add -mpegts_flags +initial_discontinuity to get rid of the corrupt
> packet messages. This may or
> >>> may not get rid of the glitches.
> >>>
> >>> Regards,
> >>> Gyan
> >>
> >> Hey, Gyan,
> >>
> >> No effect whatsoever. Same message, same glitches.
> >
> > I forgot to mention, it has to be added to the trim commands.
>
> Do you mean to '-ss' & '-to'?
>
> Oh, no, no. Even if it worked, what would I do with all the bad PTSs and
> audio frames left dangling.
>
> No, '-ss' & '-to' are disasters.
>

I usually do not chime in on your emails where people flame you for writing
a book or being a troll because I usually learn something from the
discussions.

Do you have any reaction at all to the news that decimal PTSs & DTSs are
> being truncated up? Did you
> know that? Why do you think it was done?
>

Lately you have not posted one example clip or logs from anything you have
asked questions about.  You're being vague and Gyan just gave you something
to try and you refused to try with just more questions.  If it not apparent
to you yet, the ffmpeg code behaves differently depending on the switches,
the options beneath the switches as well as the order of the switches.  You
keep wanting to recreate the atom bomb where everything is required to be
in precision order with precision timing.  Ffmpeg as stated by the
developers has been cobbled together over the years to accommodate multiple
containers and codecs inside them, yet for the last 2 weeks, you want it to
be an atom bomb, where the order of switches and codes and their options as
well as audio/video modifiers need to be used to accommodate your desired
end results "has" to be defined somewhere, then I suggest you quit talking
about your 50 year old programming of firmware and learn C so you can read
the code and answer your own questions.

Like I said, I used to be interested, but lately you are being obtuse and
lazy - wanting someone else to teach you when you should be teaching
yourself.

Sorry, but I find this list valuable and you are starting to put a bad
taste in my mouth.  And Reindl and Paul - I don't need any smart ass
comments.  Reindal, I think that we are almost on all the same lists and
everyone hates you, and Paul, you're ego would be bad in certain circles of
technology people.

--Mark.
>
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Re: [FFmpeg-user] The unsubscribe process

2023-11-29 Thread Greg Oliver
On Wed, Nov 29, 2023 at 11:56 AM Carl Zwanzig  wrote:

> On 11/29/2023 9:18 AM, David Bernat wrote:
> > You misinterpreted the response. I have emailed the correct unsubscribe
> > link several times.
> (emailed a link??? what does that mean?)
>
> In the footer of every message-
>  > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>  >
>  > To unsubscribe, visit link above, or email
>  > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
> Dropping "Please unsubscribe" in an existing message thread won't do it,
> you
> have to send the correct subject ("unsubscribe") to the correct address
> (ffmpeg-user-requ...@ffmpeg.org).
>
> Or visit the link above, enter your email, press "unsub", and wait for the
> confirmation email.
>
> z!
>

I was a little late - just sent him the same page privately :)
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Re: [FFmpeg-user] Seeking Efficient Video Redaction Using FFMPEG with Limited Memory Usage

2023-11-08 Thread Greg Oliver
On Mon, Nov 6, 2023 at 11:13 AM sreekar manchukonda
 wrote:
>
> Hello,
>
> I'm seeking help on how to utilize ffmpeg to apply an extensive number of
> filters—specifically over 2000, each filter including a crop, boxblur, and
> overlay—to a video over an hour long. The intention is to redact certain
> sections of the input at various times without consuming excessive memory.
>
> Could you advise if there's a method to manage thread pooling to limit the
> number of concurrent threads to prevent memory overflow?
>
> Also, I'm looking for a strategy to process these threads in a sequence,
> incorporating the output directly into the final video, while periodically
> clearing the memory utilized by intermediate filter operations. Presently,
> memory is only freed after the complete processing concludes.

Plex Media Server does this better than I have seen elsewhere.  It
takes an entire video with audio and splts it up into time sections
totranscode segments before plaback.  This keeps memory and disk usage
low.  You can probably get the source they use in the OpenPlex project
if you want to look at it.

> I would greatly appreciate an efficient approach to this.
>
> Thank you.
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Re: [FFmpeg-user] missing meta data in ffprobe output from mp4 format

2023-10-20 Thread Greg Oliver
On Fri, Oct 20, 2023 at 3:57 PM Ulf Zibis  wrote:
>
>
> Am 20.10.23 um 20:14 schrieb Paul B Mahol:
> > On Fri, Oct 20, 2023 at 7:19 PM Ulf Zibis  wrote:
> >
> >> Hi,
> >>
> >> in my last post "Codec 0x21000 is not in the full list."
> >> you can see, that also the included metadata is mostly missing in the
> >> output of ffprobe. Why?
> >>
> >> When I use the same command and the same metadata file for mkv output,
> >> beside this message all is fine:
> >>
> >>
> >>
> >> $ ffmpeg -i König_de.meta -t 3:00 -i König_de.mkv -t 3:00 -i König_it.mkv
> >> -map_metadata:s:v 0:s:0 -map_metadata:s:a:0 0:s:0 -map_metadata:s:a:1 0:s:1
> >> -map_metadata:s:v 1:s:v -map_metadata:s:a:0 1:s:a -map_metadata:s:a:1 2:s:a
> >> -map_metadata 0 -map_metadata 1 -map 1 -map 2:a -c copy -cues_to_front true
> >> König_de_3_c.mkv
> >> ffmpeg version 5.1.1-1ubuntu2.1 Copyright (c) 2000-2022 the FFmpeg
> >> developers
> >>
> > That version is too  old.
>
> Same problem with version 6 ...

I have never used ffprobe before, but I have seen countless messages
on this left with regards to it not being closely maintained and
recommendations to use other software to query media information.
Possibly try mediainfo of cvlc, etc and see if you have different
results...?

> $ ffmpeg -t 3:00 -i König_de.mkv -t 3:00 -i König_it.mkv -t 3:00 -i 
> König_fr.mkv -t 3:00 -i König_en.mkv \
> -i König_de.meta -map_metadata 4 \
> -map_metadata:s:v 4:s:0 -map_metadata:s:a:0 4:s:0 -map_metadata:s:a:1 4:s:1 
> -map_metadata:s:a:2 4:s:2 -map_metadata:s:a:3 4:s:3 \
> -map_metadata:s:v 0:s:v -map_metadata:s:a:0 0:s:a -map_metadata:s:a:1 1:s:a 
> -map_metadata:s:a:2 2:s:a -map_metadata:s:a:3 3:s:a \
> -map 0 -map 1:a -map 2:a -map 3:a -c copy -movflags +faststart 
> König_de_3_c.mp4
> ffmpeg version 6.0-0ubuntu1~22.04.sav3 Copyright (c) 2000-2023 the FFmpeg 
> developers
>built with gcc 11 (Ubuntu 11.4.0-1ubuntu1~22.04)
>configuration: --prefix=/usr --extra-version='0ubuntu1~22.04.sav3' 
> --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu 
> --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl 
> --disable-stripping --enable-amf --enable-gnutls --enable-ladspa 
> --enable-lcms2 --enable-libaom --enable-libass --enable-libbluray 
> --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 
> --enable-libdav1d --enable-libflite --enable-libfontconfig 
> --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm 
> --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg 
> --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq 
> --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
> --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora 
> --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx 
> --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid 
> --enable-libzimg
> --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal 
> --enable-opencl --enable-opengl --enable-sdl2 --enable-sndio 
> --enable-pocketsphinx --enable-librsvg --enable-avisynth --enable-libilbc 
> --enable-libjxl --enable-librist --enable-vapoursynth --enable-libvmaf 
> --enable-crystalhd --enable-libmfx --enable-libsvthevc --enable-libsvtav1 
> --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint 
> --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e 
> --enable-shared
>WARNING: library configuration mismatch
>avcodec configuration: --prefix=/usr 
> --extra-version='0ubuntu1~22.04.sav3' --toolchain=hardened 
> --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu 
> --arch=amd64 --enable-gpl --disable-stripping --enable-amf --enable-gnutls 
> --enable-ladspa --enable-lcms2 --enable-libaom --enable-libass 
> --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio 
> --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig 
> --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm 
> --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg 
> --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq 
> --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
> --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora 
> --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx 
> --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid
> --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx 
> --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-sndio 
> --enable-pocketsphinx --enable-librsvg --enable-avisynth --enable-libilbc 
> --enable-libjxl --enable-librist --enable-vapoursynth --enable-libvmaf 
> --enable-crystalhd --enable-libmfx --enable-libsvthevc --enable-libsvtav1 
> --enable-libdc1394 --enable-libdrm --enable-libiec61883 

Re: [FFmpeg-user] Ubuntu 22.04 audio capture issue

2022-11-30 Thread Greg Oliver
Top Posting On Purpose (with camelCase)

Could you please just stop the conversation here.  It is time consuming...

On Wed, Nov 30, 2022 at 10:08 AM Dave Blanchard  wrote:
>
>
> > I'm well-versed in "1984" and "Brave New World".  "Animal Farm" is still
> > on my to-watch and to-read lists.  I'm also well versed in the fiction
> > that is the Bible.
>
> You certainly *believe* you are, of course.
>
> > I recommend people read through the website nobeliefs.com,
>
> Yeah, I did that sort of thing decades ago also, when I was a teenager and 
> the internet was just getting started. I was the most notorious atheist in my 
> high school. I was just as ignorant then as you are today. And yes, just as 
> arrogant--like I knew something everyone else didn't. I didn't.
>
> A quote posted a few days ago from one of the other slow learners, who has 
> taken longer than I did to reach a higher level of understanding:
>
> "I don't know if it is actually in the Bible, but a lot of religious 
> Christians say that the End Times will involve the truth being accepted as 
> lies and vice-versa. One thing is certain.  Millions and millions of people 
> are under some kind of spell.  It is quite disturbing.  Whether you want to 
> call it mass formation psychosis or under the spell of evil doesn't really 
> matter.  It is real."
>
> I've been reading more and more such quotes recently, in a place where very 
> intelligent people gather to engage in high level discussions about important 
> things; it's usually along the lines of, "I never used to be a believer, but 
> with this agenda of what can only be described as absolute darkness and evil 
> descending on the world, I'm beginning to."
>
> A couple relevant quotes from a similar discussion:
>
> "Keep people on the edge with one crisis and problem and worry after another; 
> it wears down a nation.  We are now in such an era.  It is unrelenting and 
> evil!"
>
> "Astute observation. It is psychological warfare. I know a few people who've 
> checked out from the news entirely, but you can't escape a societal illness. 
> It's everywhere. Wearing down a nation and a people. People are very unhappy."
>
> > Anyway, to echo Carl Zwanzig's response, you are veering way, way off topic 
> > for the ffmpeg list.
>
> That's OK. There are certain readers out there who are glad we did veer off 
> topic for a moment.
>
> Later down the road, you WILL become much more familiar with the facts and 
> dangers I've warned about. The only question is WHEN you will learn these 
> things, and what kind of trouble you might be in at the time you do.
>
> --
> Dave Blanchard 
>
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Re: [FFmpeg-user] hls playlist got wiped out when ffmpeg runs as systemd child process

2022-07-08 Thread Greg Oliver
On Fri, Jul 8, 2022 at 5:25 AM jb  wrote:

> Hello everybody,
>
> I have a control program which running as a systemd service. This
> program spawns ffmpeg and that is writing to a HLS playlist. I use this
> flags:
>
> -hls_flags append_list+delete_segments+omit_endlist+program_date_time
>
> The interesting thing is, when I stop the systemd process my HLS
> playlist got wiped out and when I start the service again ffmpeg will
> not continue the playlist, instead it writes a new one.
>
> When I run my control program directly from shell, I don't have this
> behavior. I can stop the program and start it again and the HLS playlist
> will continue.
>
> Does anybody notice this different behaviors? And has an idea why this
> is happen and what I can do here?
>

systemd is not stateful - unless you create stateful files with your
exec{pre,post,start} stanzas, it will get overwritten every time.



> I use the current ffmpeg version from master.
>
>
> Regards
>
> Jonathan
>
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Re: [FFmpeg-user] show an image with FFplay for 5 seconds

2022-07-01 Thread Greg Oliver
On Fri, Jul 1, 2022 at 9:03 AM Michael Koch 
wrote:

> Hello,
>
> is it possible to show an image with FFplay for 5 seconds, and then
> exit? I did try this command in a Windows batch file, but it doesn't
> stop after 5 seconds. I drag and drop the image on the batch file:
>
> ffplay -autoexit -loop 0 -t 5 %1
>
> Michael
>

Definitely not a bacth file expert (have not run Windows sine the 90's),
but under linux, you could start the video in a background thread, then a
while loop checking the time.  After the time you wish is elapsed, exit the
loop, which in turn would stop any child processes as well (the ffplay
process).  I assum MS - especially now that they have a bash emulator, etc
can do this as well.
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Re: [FFmpeg-user] Change device setting from CLI?

2022-04-12 Thread Greg Oliver
On Tue, Apr 12, 2022 at 9:34 PM Greg Oliver  wrote:

> On Mon, Apr 11, 2022 at 11:28 AM Steve Russell  wrote:
>
>> Oh! Of couse... I hadn't considered that... I suspect you're right. Oh
>> well, I've learned something, thanks.
>>
>> I'll just have to live with tweaking the GUI.
>>
>> Regards, Steve
>>
>
> Practically every video capture device uses the video4linux2 framework.
> When you find the kernel module responsible for that camera, you can modify
> it's parameters so it simply "just works" everywhere the same way.
>
> https://elinux.org/images/8/89/V4l2-frameworks_0.pdf
>
> -Greg
>

Ahhh, nm - I see you are using Windows..
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Re: [FFmpeg-user] Change device setting from CLI?

2022-04-12 Thread Greg Oliver
On Mon, Apr 11, 2022 at 11:28 AM Steve Russell  wrote:

> Oh! Of couse... I hadn't considered that... I suspect you're right. Oh
> well, I've learned something, thanks.
>
> I'll just have to live with tweaking the GUI.
>
> Regards, Steve
>

Practically every video capture device uses the video4linux2 framework.
When you find the kernel module responsible for that camera, you can modify
it's parameters so it simply "just works" everywhere the same way.

https://elinux.org/images/8/89/V4l2-frameworks_0.pdf

-Greg
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Re: [FFmpeg-user] HLS to MCAST

2020-12-03 Thread Greg Oliver
On Thu, Dec 3, 2020 at 4:45 AM Максим Молодцов 
wrote:

> Hello,
>
> I've faced with a strange issue. When I try to get an HLS stream and
> convert it to multicast, like this:
>
> ffmpeg -re -timeout 6000 -m3u8_hold_counters 5 -i
> http://xxx/HLS/HD.m3u8
> > -codec:v copy -codec:a copy -f mpegts 'udp://
> 127.0.0.1:1234?pkt_size=1316'
>

I do not know how to solve your issue, only that 127.0.0.1 will never be
multicast - multicast is (only) 224.0.0.0/4 address space in IPv4



> everything works fine for several hours:
>
> [http @ 0x25a6280] Opening 'http://xxx/HLS/HD.m3u8' for reading
> > [hls @ 0x2413540] Skip ('#EXT-X-VERSION:3')
> > [hls @ 0x2413540] Skip ('#EXT-X-ALLOW-CACHE:NO')
> > [http @ 0x2582800] Opening 'http://xxx/HLS/HD2.ts' for reading
> > frame=191103 fps= 25 q=-1.0 size= 5083538kB time=02:07:24.07
> > bitrate=5447.9kbits/s speed=   1x
> > frame=191114 fps= 25 q=-1.0 size= 5083940kB time=02:07:24.51
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191127 fps= 25 q=-1.0 size= 5084228kB time=02:07:25.03
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191139 fps= 25 q=-1.0 size= 5084568kB time=02:07:25.51
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191151 fps= 25 q=-1.0 size= 5084895kB time=02:07:25.99
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191165 fps= 25 q=-1.0 size= 5085242kB time=02:07:26.55
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191179 fps= 25 q=-1.0 size= 5085529kB time=02:07:27.11
> > bitrate=5447.9kbits/s speed=   1x
> > frame=191192 fps= 25 q=-1.0 size= 5085904kB time=02:07:27.63
> > bitrate=5447.9kbits/s speed=   1x
> > frame=191202 fps= 25 q=-1.0 size= 5086246kB time=02:07:28.03
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191215 fps= 25 q=-1.0 size= 5086591kB time=02:07:28.55
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191227 fps= 25 q=-1.0 size= 5086912kB time=02:07:29.03
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191239 fps= 25 q=-1.0 size= 5087258kB time=02:07:29.51
> > bitrate=5448.0kbits/s speed=   1x
> > [http @ 0x25a6280] Opening 'http://xxx/HLS/HD.m3u8' for reading
> > [hls @ 0x2413540] Skip ('#EXT-X-VERSION:3')
> > [hls @ 0x2413540] Skip ('#EXT-X-ALLOW-CACHE:NO')
> > [http @ 0x25684c0] Opening 'http://xxx/HLS/HD3.ts' for reading
> > frame=191252 fps= 25 q=-1.0 size= 5087605kB time=02:07:30.03
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191266 fps= 25 q=-1.0 size= 5087919kB time=02:07:30.59
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191279 fps= 25 q=-1.0 size= 5088265kB time=02:07:31.11
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191290 fps= 25 q=-1.0 size= 5088586kB time=02:07:31.55
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191303 fps= 25 q=-1.0 size= 5088931kB time=02:07:32.07
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191315 fps= 25 q=-1.0 size= 5089249kB time=02:07:32.55
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191327 fps= 25 q=-1.0 size= 5089594kB time=02:07:33.03
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191341 fps= 25 q=-1.0 size= 5089967kB time=02:07:33.59
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191354 fps= 25 q=-1.0 size= 5090260kB time=02:07:34.11
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191367 fps= 25 q=-1.0 size= 5090612kB time=02:07:34.63
> > bitrate=5448.0kbits/s speed=   1x
> > frame=191380 fps= 25 q=-1.0 size= 5090923kB time=02:07:35.15
> > bitrate=5447.9kbits/s speed=   1x
> > frame=191393 fps= 25 q=-1.0 size= 5091242kB time=02:07:35.67
> > bitrate=5447.9kbits/s speed=   1x
> > [http @ 0x25a6280] Opening 'http://xxx/HLS/HD.m3u8' for reading
> > [hls @ 0x2413540] Skip ('#EXT-X-VERSION:3')
> > [hls @ 0x2413540] Skip ('#EXT-X-ALLOW-CACHE:NO')
> > ...
> >
>
> But after a few hours, the multicast monitoring tool gives out:
>
> > Nov 07 00:51:01: INFO: Bitrate: 30287 Kbit/s
> > Nov 07 00:51:02: INFO: Bitrate: 9610 Kbit/s
> > Nov 07 00:51:03: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:04: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:05: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:06: INFO: Bitrate: 36193 Kbit/s
> > Nov 07 00:51:07: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:08: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:09: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:10: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:11: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:12: INFO: Bitrate: 12801 Kbit/s
> > Nov 07 00:51:13: INFO: Bitrate: 10757 Kbit/s
> > Nov 07 00:51:14: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:15: INFO: Bitrate: 0 Kbit/s
> > Nov 07 00:51:16: INFO: Bitrate: 0 Kbit/s
> >
>
> and in the ffmpeg log I see fewer lines between
>
> > Opening 'http://xxx/HLS/HD2.ts' for reading
> >
>
> :
>
> [http @ 0x2765200] Opening 'http://xxx/HLS/HD0.ts' for reading
> > frame=1845421 fps= 25 q=-1.0 size=49087105kB time=20:30:20.83
> > bitrate=5447.3kbits/s speed=   1x
> > [http @ 0x2888200] Opening 'http://xxx/HLS/HD.m3u8' for reading
> > [hls @ 0x275c540] Skip ('#EXT-X-VERSION:3')
> > [hls @ 0x275c540] Skip ('#EXT-X-ALLOW-CACHE:NO')
> > [http @ 0x2765200] Opening 'http://xxx/HLS/HD1.ts' for reading
> > 

Re: [FFmpeg-user] Compilation Guide / MSVC on Wiki

2020-12-02 Thread Greg Oliver
On Wed, Dec 2, 2020 at 11:43 AM Carl Zwanzig  wrote:

> On 12/2/2020 9:32 AM, hinderanyoption81 via ffmpeg-user wrote:
> > Upon reading the "Compilation Guide / MSVC on Wiki", I am confused about
> at least three issues.
> > 1. The Guide starts with a sentence "FFmpeg can be built on Windows with
> Visual Studio", and then it says:
> >
> > Finally, run: For MSVC:
> > ./configure --toolchain=msvc
> > make
> > make install
>
> Yep, the instructions aren't clear.
>
> I've successfully built ffmpeg on/for windows using msys2/mingw-w64. You
> need to install msys2, then the compiler packages, then things like (IIRC)
> automake and it's friends, then others. I can look that up later, but the
> first time was a somewhat iterative process in "Can't find ", so I
> install  and try again; eventually it all worked.
>
> Start at https://www.msys2.org/. Also, there are msys and mingw packages,
> most of the time, you need the mingw one.
>
> This is a very useful setup on a windows system.
>
>
> To answer the MSVC question- I only use it for things that aren't set up
> to
> build in mingw :).
>
> Later,
>
> z!
>

All of this makes me wonder -

A lot of folks here run Windows (fine, I get it).  Windows 10 includes a
*free* Hyper-V hypervisor.  These same people are wanting to manipulate
audio/video or they would not be here and as well not have the disk space
for a virtual guest.  The instructions and packages for Linux are available
almost everywhere, so why not someone provide directions on creating a
Hyper-V guest of Ubuntu (arguably the easiest distro for newbies to Linux)
and showing how to get it installed and add ffmpeg packages.  I have not
been on this list long, but I can say that the strife of people looking for
non-existent windows binaries is frequent even though we know there are a
few shared library build as well as (at least one I know of) static build
with no dependencies.  It just seems like a lot of work for something so
easy to bypass with another route.

That route is so much easier than compiling *anything* under Windows since
MS keeps under lock and key all of their library code for all of their
compilers and linkers, etc..

It just seems a no-brainer to me.  Much easier route to both explain and
maintain..  Setup some shared folders between host and guest, Linux can see
the files that actually reside on the Windows host and write the results
right back to the share so they show up in Windows natively.
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Re: [FFmpeg-user] Non-mandatory audio

2020-10-14 Thread Greg Oliver
On Wed, Oct 14, 2020 at 10:40 AM Will Boba  wrote:

> On 10/14/20 11:18 AM, Greg Oliver wrote:
> > sorry just realized I had to redirect stderr to get the results
> >
> > On Wed, Oct 14, 2020 at 10:17 AM Greg Oliver 
> wrote:
> >
> >> On Tue, Oct 13, 2020 at 8:50 AM Mar Andrés López 
> >> wrote:
> >>
> >>> Dear all,
> >>>
> >>>
> >>> I am using the following command to generate dash content from a file:
> >>>
> >>>
> >>> ffmpeg -i video.mp4 –r 25 -c:v libx264 -c:a aac -b:a 128k -ac 1 -ar
> 44100
> >>> -map v:0 -b:v:0 2M -map v:0 -b:v:1 145k -map v:0 -b:v:2 365k -map 0:a
> -f
> >>> dash dash.mpd
> >>>
> >>>
> >>> But sometimes I have streams with audio and sometimes not. Is there
> >>> anyway to make this command to not break if the video file does not
> >>> contains audio?
> >>>
> >>>
> >>> A lot of thanks
> >>>
> >> Assuming you are scripting this, probe the file first and check for
> >> audio.  Use a separate ffmpeg command line for eac result.
> >>
> >> ffprobe  | grep "Stream.*Audio"
> >> if [ $? -eq 0 ]; then
> >>  ffmpeg command with audio
> >> else
> >>  ffmpeg command without audio
> >> fi
> >>
> >
> > ffprobe  2>&1 | grep "Stream.*Audio"
> > if [ $? -eq 0 ]; then
> >  ffmpeg command with audio
> > else
> >  ffmpeg command without audio
> > fi
> > ___
> > ffmpeg-user mailing list
> > ffmpeg-user@ffmpeg.org
> > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> > To unsubscribe, visit link above, or email
> > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
>  From here: https://ffmpeg.org/ffmpeg-all.html
>
> "To map the video and audio streams from the first input, and using the
> trailing ?, ignore the audio mapping if no audio streams exist in the
> first input:
>
> ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT"
>
Even easier :)  Nice to know this exists.



>
> Will.
>
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Re: [FFmpeg-user] Non-mandatory audio

2020-10-14 Thread Greg Oliver
On Tue, Oct 13, 2020 at 8:50 AM Mar Andrés López 
wrote:

> Dear all,
>
>
> I am using the following command to generate dash content from a file:
>
>
> ffmpeg -i video.mp4 –r 25 -c:v libx264 -c:a aac -b:a 128k -ac 1 -ar 44100
> -map v:0 -b:v:0 2M -map v:0 -b:v:1 145k -map v:0 -b:v:2 365k -map 0:a -f
> dash dash.mpd
>
>
> But sometimes I have streams with audio and sometimes not. Is there anyway
> to make this command to not break if the video file does not contains audio?
>
>
> A lot of thanks
>

Assuming you are scripting this, probe the file first and check for audio.
Use a separate ffmpeg command line for eac result.

ffprobe  | grep "Stream.*Audio"
if [ $? -eq 0 ]; then
ffmpeg command with audio
else
ffmpeg command without audio
fi
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Re: [FFmpeg-user] Non-mandatory audio

2020-10-14 Thread Greg Oliver
sorry just realized I had to redirect stderr to get the results

On Wed, Oct 14, 2020 at 10:17 AM Greg Oliver  wrote:

> On Tue, Oct 13, 2020 at 8:50 AM Mar Andrés López 
> wrote:
>
>> Dear all,
>>
>>
>> I am using the following command to generate dash content from a file:
>>
>>
>> ffmpeg -i video.mp4 –r 25 -c:v libx264 -c:a aac -b:a 128k -ac 1 -ar 44100
>> -map v:0 -b:v:0 2M -map v:0 -b:v:1 145k -map v:0 -b:v:2 365k -map 0:a -f
>> dash dash.mpd
>>
>>
>> But sometimes I have streams with audio and sometimes not. Is there
>> anyway to make this command to not break if the video file does not
>> contains audio?
>>
>>
>> A lot of thanks
>>
>
> Assuming you are scripting this, probe the file first and check for
> audio.  Use a separate ffmpeg command line for eac result.
>
> ffprobe  | grep "Stream.*Audio"
> if [ $? -eq 0 ]; then
> ffmpeg command with audio
> else
> ffmpeg command without audio
> fi
>


ffprobe  2>&1 | grep "Stream.*Audio"
if [ $? -eq 0 ]; then
ffmpeg command with audio
else
ffmpeg command without audio
fi
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Re: [FFmpeg-user] Glossary: Nyquist

2020-10-01 Thread Greg Oliver
On Wed, Sep 30, 2020 at 6:25 PM Mark Filipak (ffmpeg) 
wrote:

> Nyquist [adjective]: 1, Reference to the Nyquist–Shannon sampling
>theorem. 2, The principle [1] that, to most faithfully reproduce an
>image at a given digital display's resolution, the samples must be
>made at or above twice the display's resolution, both horizontally &
>vertically [2].
>[1] The Nyquist principle applies to film sampling and to digital
>cameras, but, provided that resolution is unchanged, not to
>transcoding (because the transcoder inputs are already digital).
>As proved by the improved appearance of SD media made from 2K
>samples, SD mastering prior to the advent of 2K sampling (e.g.
>DVDs mastered before the advent of HD) generally ignored the
>Nyquist principle and were undersampled. HDs sampled at 2K and
>UHDs sampled at 4K are likewise undersampled.
>[2] As a convenience, the Nyquist threshold is currently (in 2020)
>specified solely by horizontal sample count rounded up to whole
>kilo-samples (2K, 4K, 8K).
>  displayNyquist threshold
>UHD 16:9-2160:  3840 x 2160 8K
> 4:3-2160:  2880 x 2160 8K
> HD 16:9-1080:  1920 x 1080 4K
> 4:3-1080:  1440 x 1080 4K
> SD  16:9-576:  1024 x 576  4K
>  4:3-576:   768 x 576  2K
> 16:9-480:   853 x 480  2K
>  4:3-480:   640 x 480  2K



Mark,

Normally I would absolutely defend your queries as they are technical and
lower level, but I would almost have to side with Bouke from post
(
bwdif filter question
)

You are trying to get free editing for your book now.  I do not agree with
that..  There are many good contributors and inquisitors (you included),
but (IMHO) you cannot solicit things like this that are grammatical rather
than technical.   I think a lot of the developers are also in the same boat
as you (sometimes) try to re-define things that are common language (even
if not accurate technically).

eg - your definition if interlaced versus interweaved..  No matter if you
are right or wrong, the concept and understanding of a majority will
prevail - no exceptions.

Please (for me at least) keep your posts here related to ffmpeg and not
trying to change the nomenclature of what exists.  We are all using the
same software, so whatever the software uses for terminology (as this list
is exactly related to), please do not interfere with that.

Take that up directly with developers and let them sort it out.

On a side note - I have yet seen one of your definitions of a technology
been held up when a developer chimes in - no hard feelings, just that
industry terminology is hard to trump :)

-Greg
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Re: [FFmpeg-user] bwdif filter question

2020-09-23 Thread Greg Oliver
On Tue, Sep 22, 2020 at 1:14 PM Mark Filipak (ffmpeg) 
wrote:

> I could use some help in my efforts. Due to my ignorance, it's taking me
> weeks to figure out things
> that should be resolved in minutes. In 5 days, I'll be 74 years old. With
> corona virus and age, I
> don't know how much longer I'll be around, but I'm sure I can help the
> ffmpeg project if the
> principals in the project will just stop sniping at me and share knowledge.
>
> Thank you for your oh-so valuable contributions. I will study AVFrame to
> see how I can use it to
> better communicate.


I must say that I am by no means an ffmpeg expert (or even an advanced
user).  I am on more mailing lists than your average person though. The
fact that "troll" keeps coming up in response to Mark's posts is quite
annoying.  He has repeatedly posted to either understand or define better
the internals of ffmpeg itself.  His posts are always well defined and
pointed - if that is hard for you who want to define it as trolling to
understand, then you should kick rocks.  I personally learn more about
ffmpeg from his posts (and therefore the responses from understandable
others) than I would otherwise from reading outdated wiki and docs.

If you feel threatened by people who desire knowledge or otherwise demand
(subtlety) proper definitions of technology - maybe you should troll
elsewhere.  I have never experienced a post on this list that maintains
otherwise.

Please quit making me read the BS that accompanies most of the intelligent
questions or contradiction responses - I find it very  educational in the
least.
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Re: [FFmpeg-user] how to control quality of subtitles

2020-06-10 Thread Greg Oliver
On Mon, May 25, 2020 at 11:58 AM Peter Münster  wrote:

> On Mon, May 25 2020, Carl Eugen Hoyos wrote:
>
> > Please provide an input sample.
>
> Here is a sample, this time without corruption:
>
> http://pmrb.free.fr/tmp/ffmpeg-test/sample.mkv
>
> --
>Peter
>

Sorry to chime in late -

Samsungs are notorious for display info from a "Computer" very badly unless
you change the name of the input on the TV to "PC" - yes this includes HDMI
inputs as well.

Have you tried that?  All of our Samsungs have a dedicated input already
named PC, but you can name as many as you like to PC.

Info ::
https://www.samsung.com/za/support/tv-audio-video/hdmi-pc-connection-text-is-fuzzy-image-is-not-clear/
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Re: [FFmpeg-user] Muxing multiple files and concatenating those outputs

2020-04-22 Thread Greg Oliver
On Tue, Apr 21, 2020 at 9:40 PM Edward Park  wrote:

> Hi,
>
> > I ended up just keeping it as a script and getting the developers to
> > incorporate it into our code.  Thanks though!  Since it is a script, a
> > single bash call is not too intrusive.  If we had some C programmers, I
> > would have them use the ffmpeg libraries directly, but they are all just
> > java heads..
>
> So is the problem basically solved now? If I had known this zombie
> apocalypse-like status quo was going to maintain for this long I would have
> recommended seeing if your VAR or sales rep if you have an account with the
> vendor for your call management system to see if you can get a license for
> doing this automatically (call recording's probably a feature? Just not
> activated I'm thiinking?)
>
> Regards,
> Ted Park


We are the call recording vendor :)

Yes, it is solved (basically just kept it like it was prior to y email to
the list).  Thanks everyone for their input.
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Re: [FFmpeg-user] Muxing multiple files and concatenating those outputs

2020-04-21 Thread Greg Oliver
On Thu, Apr 16, 2020 at 10:19 AM Nicolas George  wrote:

> Greg Oliver (12020-04-14):
> > We currently use ffmpeg to combine 2 streams (telephony codecs) into
> > mp3/wav/flac, etc..  This is no problem - we use:
> >
> > ffmpeg -ar 8000 -y -f mulaw -i calls/example.g711a.ulaw -ar 8000 -f mulaw
> > -i calls/example.g711b.ulaw -filter_complex "[0:a][1:a]amerge[aout]" -map
> > "[aout]" output.wav
> >
> > If we have multiple legs of a recording, we just use linux `cat' to
> > concatenate them prior to ffmpeg processing (we have raw audio frames
> > without headers, etc, so this works fine).
> >
> > Due to Covid, almost all of our customers have moved their call centers
> to
> > employee homes, so we have mixed codec calls and the necessity has arose
> to
> > generate audio files accordingly.  It is easily accomplished with
> multiple
> > temporary files, but I would like to clean it up and have been
> unsuccessful
> > in coming up with a single string to accomplish it.
> >
> > Example using these files (suffix denotes codec - a and b are each side
> of
> > a call):
> >
> > 18_17_248.g729a
> > 18_17_248.g729b
> > 19_18_440.g711a
> > 19_18_440.g711b
> > 20_01_886.g729a
> > 20_01_886.g729b
> >
> > Current method to concatenate and transcode to wav putting caller on left
> > and callee on right:
> >
> > ffmpeg -f g729 -i 18_17_248.g729a -f g729 -i 18_17_248.g729b
> > -filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_1.wav
> >
> > ffmpeg -ar 8000 -y -f mulaw -i 19_18_440.g711a -ar 8000 -f mulaw -i
> > 19_18_440.g711b -filter_complex "[0:a][1:a]amerge[aout]" -map "[aout]"
> > out_2.wav
> >
> > ffmpeg -f g729 -i 20_01_886.g729a -f g729 -i 20_01_886.g729b
> > -filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_3.wav
> >
> > ffmpeg out_1.wav out_2.wav out_3.wav final.wav
> >
> > My goal is to syntactically accomplish this with one ffmpeg string (not
> > pipes, ; or && bash syntax).
> >
> > I would appreciate any insight - I have tried everything I can find /
> think
> > of without success.
>
> I suspect what you need is the concat filter. But you were not clear
> enough about what exactly you are trying to achieve. Maybe make diagram?
>
> Regards,
>
> --
>   Nicolas George


I ended up just keeping it as a script and getting the developers to
incorporate it into our code.  Thanks though!  Since it is a script, a
single bash call is not too intrusive.  If we had some C programmers, I
would have them use the ffmpeg libraries directly, but they are all just
java heads..

-Greg
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Re: [FFmpeg-user] ffplay options for setting audio volume and locating player windows on the screen

2020-04-15 Thread Greg Oliver
On Tue, Apr 14, 2020 at 8:46 PM Michael Glenn Williams <
michael.glenn.willi...@smartblondeco.com> wrote:

> Hey Ted!
>
> Thank you for responding. Regarding -volume, i'm just putting 30
> without the %.
> Regarding -top and -left, my version of ffplay (the latest MAC OSX download
> from the website)
> Doesn't seem to know these options.
>

Have you tried the good ole tried and true X / XQuartz -geometry options?
I have not used OSX in a very long time (since the switch to Quartz), so
there may be some differences between the old Xorg and XFree nomenclature,
but I imagine those types of options would still be there for initial
window placements and sizes.



> I'm confused why generic options like these would vary from
> build/distribution to build?
>
>
>
>
> On Tue, Apr 14, 2020 at 5:31 PM Ted Park  wrote:
>
> > Hi,
> >
> > > I tried the -w and -h options, which were not found by ffplay. The -x
> and
> > > -y options seem to be the options that set the width and heoght, not
> the
> > > location on the screen. Can anyone tell me how to position the windows
> in
> > > different places on the screen?
> > >
> > > I also tried the -volume option to start out at 30% volume, but that
> > option
> > > also wasn't found.
> > I'm not sure if it's the same across all libraries/OS's, but -left and
> > -top set the distance from left and top of screen respectively, for me.
> And
> > when you specify volume, try not including the "%" sign.
> >
> > Regards,
> > Ted Park
> >
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Re: [FFmpeg-user] Muxing multiple files and concatenating those outputs

2020-04-15 Thread Greg Oliver
On Wed, Apr 15, 2020 at 5:43 AM Ted Park  wrote:

> Hi,
>
> > Yeah, when we record the calls, the directory structure preceeding those
> > names is /MM/DD/HH and the filenames are MIN_SEC_MSEC.codec(side)
>
>
> I mean some phones specifically put g729a, and I assume it's the same for
> g729b. So I started  imagining files ending all over the place g729aa,
> g729ba, etc.
>

Ahhh, I see what you are referring to now :)

g729b is just an annex to a that only allows for more header enhancements -
the raw codec audio is the same though, so we just write g729 for both.



> > I was really looking for just syntax to group commands, so I could use
> the
> > merge filter output as direct input for the concatenation.
>
> If you mean the afifo I inserted, you could probably get rid of those if
> the machine's fast enough, or the calls short enough. Or a different
> approach might be padding each file in the same call or using the cue
> filter, and mixing them all. But grouping commands isn't really a thing
> that ffmpeg does, except for things like image sequences, preparing
> segmented delivery media.
>

I see the restraints I am up against now and I think what I am trying is
just not possible, so I will take the path of least resistance and keep it
multi-lined.  To clarify, the reason I need to merge channels prior is that
the 2 sides do not always exactly line up (there almost always is a
fraction of a second less on one side of the packet stream) and it causes
conversation sync/skew if they are concatenated either prior or
simultaneously with the merging.



> > The man page and web searches I tried came up empty, but I figured
> someone may know some magic sauce I could not find.
>
>
> Maybe you've been looking for the wrong terms? It sounds to me like what
> you are looking for is closer to shell features, like parameter
> substitution, filename generation, etc.
>

Yeah, I am currently using mktemp, etc..  I was just hoping to simplify it
because IO is the limiting factor when it comes to call recording, so
minimizing it is always a prioriry for me.



> If portability is not an issue, some shells have more features than
> others, though there's a different learning curve to each one.
>
> Or there's always xargs. Yeah probably xargs. I can't really tell but the
> associated filenames seem pretty much arbitrary. Do you parse the date/time
> in the filename to find which ones to put together? Or is there a call log
> to reference?
>
> Regards,
> Ted Park
>

Thanks again for all of your input Ted.

-Greg
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Re: [FFmpeg-user] Muxing multiple files and concatenating those outputs

2020-04-14 Thread Greg Oliver
On Tue, Apr 14, 2020 at 3:30 PM Ted Park  wrote:

> Hi,
>
> > Example using these files (suffix denotes codec - a and b are each side
> of
> > a call):
> >
> > 18_17_248.g729a
> > 18_17_248.g729b
> > 19_18_440.g711a
> > 19_18_440.g711b
> > 20_01_886.g729a
> > 20_01_886.g729b
> >
> That is really confusing...
>

Yeah, when we record the calls, the directory structure preceeding those
names is /MM/DD/HH and the filenames are MIN_SEC_MSEC.codec(side)



> > Current method to concatenate and transcode to wav putting caller on left
> > and callee on right:
> >
> > ffmpeg -f g729 -i 18_17_248.g729a -f g729 -i 18_17_248.g729b
> > -filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_1.wav
> >
> > ffmpeg -ar 8000 -y -f mulaw -i 19_18_440.g711a -ar 8000 -f mulaw -i
> > 19_18_440.g711b -filter_complex "[0:a][1:a]amerge[aout]" -map "[aout]"
> > out_2.wav
> >
> > ffmpeg -f g729 -i 20_01_886.g729a -f g729 -i 20_01_886.g729b
> > -filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_3.wav
> >
> > ffmpeg out_1.wav out_2.wav out_3.wav final.wav
> >
> > My goal is to syntactically accomplish this with one ffmpeg string (not
> > pipes, ; or && bash syntax).
> >
> > I would appreciate any insight - I have tried everything I can find /
> think
> > of without success.
> >
> The final command is supposed to concatenate the out_?.wav files into the
> final.wav file right?
>
> I think the way you have them written, it makes it easy to combine it into
> one command, though I have to wonder what benefit (if any) you seek to get
> from it. Because, if you're scripting it, it really shouldn't matter, and
> if you're doing it by hand, doesn't it make one big command that has to be
> repeated if it fails? Anyhow,
>
> > ffmpeg -f g729 -i 18_17_248.g729a -f g729 -i 18_17_248.g729b -ar 8000 -y
> -f mulaw -i 19_18_440.g711a -ar 8000 -f mulaw -i
> > 19_18_440.g711b -f g729 -i 20_01_886.g729a -f g729 -i 20_01_886.g729b
> -filter_complex
> '[0][1]amerge,afifo[l1];[2][3]amerge,afifo[l2];[4][5]amerge,afifo[l3];[l1][l2][l3]concat=v=0:a=1:n=3'
> 
>
> I think this will do what you want, the individual input pads to amerge
> and/or fifo might not be needed.
>

Yeah, the only issue is that if there are 10 legs of a call (think patient
calling a hospital and asking for neurology, getting the neurology
receptionist and then being transferred to a doctor who does not answer and
rings back to the receptionist who does not answer and then finally
arriving back at the main switchboard and the cycle starts over).  We may
have quite a few legs, so we must mux each leg prior to the concatenation.

I was really looking for just syntax to group commands, so I could use the
merge filter output as direct input for the concatenation.  The man page
and web searches I tried came up empty, but I figured someone may know some
magic sauce I could not find.

Thanks!



> Regards,
> Ted Park
>
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[FFmpeg-user] Muxing multiple files and concatenating those outputs

2020-04-14 Thread Greg Oliver
Hi,

We currently use ffmpeg to combine 2 streams (telephony codecs) into
mp3/wav/flac, etc..  This is no problem - we use:

ffmpeg -ar 8000 -y -f mulaw -i calls/example.g711a.ulaw -ar 8000 -f mulaw
-i calls/example.g711b.ulaw -filter_complex "[0:a][1:a]amerge[aout]" -map
"[aout]" output.wav

If we have multiple legs of a recording, we just use linux `cat' to
concatenate them prior to ffmpeg processing (we have raw audio frames
without headers, etc, so this works fine).

Due to Covid, almost all of our customers have moved their call centers to
employee homes, so we have mixed codec calls and the necessity has arose to
generate audio files accordingly.  It is easily accomplished with multiple
temporary files, but I would like to clean it up and have been unsuccessful
in coming up with a single string to accomplish it.

Example using these files (suffix denotes codec - a and b are each side of
a call):

18_17_248.g729a
18_17_248.g729b
19_18_440.g711a
19_18_440.g711b
20_01_886.g729a
20_01_886.g729b

Current method to concatenate and transcode to wav putting caller on left
and callee on right:

ffmpeg -f g729 -i 18_17_248.g729a -f g729 -i 18_17_248.g729b
-filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_1.wav

ffmpeg -ar 8000 -y -f mulaw -i 19_18_440.g711a -ar 8000 -f mulaw -i
19_18_440.g711b -filter_complex "[0:a][1:a]amerge[aout]" -map "[aout]"
out_2.wav

ffmpeg -f g729 -i 20_01_886.g729a -f g729 -i 20_01_886.g729b
-filter_complex "[0:a][1:]amerge[aout]" -map "[aout]" out_3.wav

ffmpeg out_1.wav out_2.wav out_3.wav final.wav

My goal is to syntactically accomplish this with one ffmpeg string (not
pipes, ; or && bash syntax).

I would appreciate any insight - I have tried everything I can find / think
of without success.

TiA,

Greg
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