Re: [FFmpeg-user] Report non Compliance with License

[Reindl Harald]

Am 29.01.2017 um 22:54 schrieb Carl Eugen Hoyos:

2017-01-29 21:02 GMT+01:00 Reindl Harald :


Am 29.01.2017 um 20:53 schrieb Phil Rhodes:


 Make sure the source code corresponds exactly to the library
binaries you are distributing.


How is this to be proven?


it's impossible to prove it


This is not generally correct:
The Debian builds are - for example - binary-reproducible.


but these are no debian builds and without repeatable builds you can't 
prove it nor proves a binary with a different hash build from the 
sources that these where not used for the shipped binary


without the 100% exactly build environment including all versions of 
involved software and libraries you can't prove it



but it's possible to prove the opposite in some cases


It's usually easy: As reported, Todd, Michael & James, Inc. and Video Surgeon
are violating the copyrights of the FFmpeg developers


as said: in some cases, in a closed appliance you would even have a 
problem to prove that ffmpeg exists there at all



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[FFmpeg-user] Question about mp3 bit rate mode

[桃源老師]

Hello ffmpeg experts...

I have question about mp3 bit rate mode.  

When I run the following command, the resulting duration.mp3 is VBR.  I have 
confirmed it's bit rate mode with mediainfo, mpg123, and SinkuSuperLite (on 
Windows).

$ ffmpeg -f lavfi -i sine=d=300 duration.mp3
ffmpeg version N-83152-gf7e9275 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --prefix=/Volumes/ffmpeg_compile --pkg-config-flags=--static 
--disable-ffserver --enable-gpl --enable-version3 --enable-nonfree 
--enable-libmp3lame --enable-libfdk-aac --enable-libvo-amrwbenc 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvorbis 
--enable-libspeex --enable-libopus --enable-libgsm --enable-libtwolame 
--enable-libsoxr --enable-libwavpack --enable-libmodplug --enable-libopenjpeg 
--enable-libwebp --enable-libtheora --enable-libx264 --enable-libopenh264 
--enable-libx265 --enable-libxvid --enable-libvpx --enable-libxavs 
--enable-libfreetype --enable-fontconfig --enable-libfribidi --enable-libass 
--enable-libbluray --enable-libvidstab --enable-libschroedinger 
--enable-libsnappy --enable-libzmq
  libavutil  55. 43.100 / 55. 43.100
  libavcodec 57. 74.100 / 57. 74.100
  libavformat57. 62.100 / 57. 62.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 69.100 /  6. 69.100
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
Input #0, lavfi, from 'sine=d=300':
  Duration: N/A, start: 0.00, bitrate: 705 kb/s
Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
Output #0, mp3, to 'duration.mp3':
  Metadata:
TSSE: Lavf57.62.100
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
Metadata:
  encoder : Lavc57.74.100 libmp3lame
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 128x
video:0kB audio:2344kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 0.009457%

And when I run the following command, the resulting duration.mp3 is CBR.  
Confirmed with same program as above.

$ ffmpeg -f lavfi -i sine=d=300 -write_xing 0 duration.mp3
ffmpeg version N-83152-gf7e9275 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --prefix=/Volumes/ffmpeg_compile --pkg-config-flags=--static 
--disable-ffserver --enable-gpl --enable-version3 --enable-nonfree 
--enable-libmp3lame --enable-libfdk-aac --enable-libvo-amrwbenc 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvorbis 
--enable-libspeex --enable-libopus --enable-libgsm --enable-libtwolame 
--enable-libsoxr --enable-libwavpack --enable-libmodplug --enable-libopenjpeg 
--enable-libwebp --enable-libtheora --enable-libx264 --enable-libopenh264 
--enable-libx265 --enable-libxvid --enable-libvpx --enable-libxavs 
--enable-libfreetype --enable-fontconfig --enable-libfribidi --enable-libass 
--enable-libbluray --enable-libvidstab --enable-libschroedinger 
--enable-libsnappy --enable-libzmq
  libavutil  55. 43.100 / 55. 43.100
  libavcodec 57. 74.100 / 57. 74.100
  libavformat57. 62.100 / 57. 62.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 69.100 /  6. 69.100
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
Input #0, lavfi, from 'sine=d=300':
  Duration: N/A, start: 0.00, bitrate: 705 kb/s
Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
File 'duration.mp3' already exists. Overwrite ? [y/N] y
Output #0, mp3, to 'duration.mp3':
  Metadata:
TSSE: Lavf57.62.100
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
Metadata:
  encoder : Lavc57.74.100 libmp3lame
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 134x
video:0kB audio:2344kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 0.001875%


I'd like to know why "-write_xing 0" changes bit rate mode of mp3.


Best Regards,


// Miya aka. TougenRoushi



smime.p7s
Description: S/MIME cryptographic signature
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[FFmpeg-user] Weird crackling sound when muxing with an existing WAV file

[Michael Heuberger]

Hello guys

It's me Michael again, having another ffmpeg problem. Since I am Deaf I
cannot fine tune nor adjust audio stuff very well and would appreciate
your help a lot.

It's for the cool www.videomail.io app where Deaf people can exchange
stuff in sign language. But the sound part needs improvement. It works
but there is a nasty crackling sound my hearing friends say. Feel free
to test it yourself on that site.

Here's an example ffmpeg command how I mux a WAV file + encode the video
based on images on server side at the same time. With a long output, all
on this Gist:
https://gist.github.com/binarykitchen/fc6c73aca72881f76376fb274d82cd70

The WAV file itself is assembled on server side from PCM files sent from
the client (during recording). Plus stretched with Rubberband before it
is being muxed with the video file. You see, it's pretty sophisticated
and works well. Just the crackling sound is a problem.

Any advice, tips or even criticism would be very very welcome

Many thanks
Michael

-- 

Binary Kitchen
Michael Heuberger
1/33 Parrish Road
Sandringham
Auckland 1025
(New Zealand)

Mobile (text only) ...  +64 21 261 89 81
Email   mich...@binarykitchen.com
Website ..  http://www.binarykitchen.com

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Re: [FFmpeg-user] Convert MPEG-PS progressive to Interlaced

[Rens Dijkshoorn]

 
>> For proper interlaced encoding the encoder needs to be
>> interlaced aware of this field structure and the possiblity
>> of temporal difference between fields in one video frame.
> 
> But even then the output would (most likely, command line
> and complete, uncut console output missing) still not be
> field-encoded.
> 

Depends what you mean by that, this command will create an
interlaced frame from 25 FPS footage but as they originate
from the same frame there is no temporal difference between
top and bottom fields.

It will not provide intra-field encoding. MPEG2 is to my
knowledge always frame base but capable to preserve
interlaced fields within the frame if flagged appropriate
as interlaced.

Rens
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Re: [FFmpeg-user] Convert MPEG-PS progressive to Interlaced

[Carl Eugen Hoyos]

2017-01-29 14:08 GMT+01:00 Rens Dijkshoorn :

> For proper interlaced encoding the encoder needs to be
> interlaced aware of this field structure and the possiblity
> of temporal difference between fields in one video frame.

But even then the output would (most likely, command line
and complete, uncut console output missing) still not be
field-encoded.

Carl Eugen
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Re: [FFmpeg-user] Static ARMEL ffmpeg build on Zero not resolving domains properly?

[Carl Eugen Hoyos]

2017-01-26 11:09 GMT+01:00 christina zou :

> I have a Pi Zero

Do you have a C compiler for this platform?
If not, I am not sure how to help, if you have
one, compilation should work out-of-the-box.

Carl Eugen
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Re: [FFmpeg-user] ERROR: libx264 not found

[Carl Eugen Hoyos]

2017-01-26 11:44 GMT+01:00 Vlad Rahmileviz :

> I'm trying to build FFMPEG with "--enable-libx264" option.
> It fails on configure
> ($ ./configure --enable-asm --enable-yasm --enable-shared --disable-static 
> --prefix=/c/ffmpeg --toolchain=msvc --arch=x86_64 --extra-cflags="-MD" 
> --enable-gpl --enable-libx264)

> ./ffconf.YAf6YOV3.c(1): fatal error C1083: Cannot open include file: 
> 'x264.h': No such file or directory

You have to install the necessary header x264.h (and the actual
library) in a place
where the compiler will find it by default or tell configure (and the
compiler) where
to find them with --extra-cflags and --extra-ldflags.

Unrelated:
The following configure options have no effect, please remove them:
--enable-asm --enable-yasm --arch=x86_64

Carl Eugen
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Re: [FFmpeg-user] Report non Compliance with License

[Carl Eugen Hoyos]

2017-01-29 21:02 GMT+01:00 Reindl Harald :
>
> Am 29.01.2017 um 20:53 schrieb Phil Rhodes:
>>>
>>>  Make sure the source code corresponds exactly to the library
>>> binaries you are distributing.
>>
>> How is this to be proven?
>
> it's impossible to prove it

This is not generally correct:
The Debian builds are - for example - binary-reproducible.

> but it's possible to prove the opposite in some cases

It's usually easy: As reported, Todd, Michael & James, Inc. and Video Surgeon
are violating the copyrights of the FFmpeg developers.

Carl Eugen
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Re: [FFmpeg-user] Report non Compliance with License

[Carl Eugen Hoyos]

2017-01-27 7:15 GMT+01:00 tracy :

> I'm attempting to contact someone with whom I can report a non
> compliance with ffmpsg's License terms.

Why?

What I mean is:
Todd, Michael & James, Inc. and Video Surgeon are violating the
copyrights of the FFmpeg developers but what should (or could)
we do about it?
Do you agree that it makes more sense to improve our software
instead of investing time to go after license violators?

I suspect nobody of the FFmpeg developers lives in Pennsylvania
(where they act from afaiu) and iirc, we even were unsuccessful
to remove violating apps from the app store;-(

Thanks for informing us!

Carl Eugen
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Re: [FFmpeg-user] Suggestions for FFmpeg: support a simpler command line for concatenating audio files with a specified overlap

[Cley Faye]

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

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

[Reindl Harald]

Am 29.01.2017 um 20:53 schrieb Phil Rhodes:

 Make sure the source code corresponds exactly to the librarybinaries you 
are distributing.

How is this to be proven?


it's impossible to prove it
but it's possible to prove the opposite in some cases
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Re: [FFmpeg-user] Report non Compliance with License

[Phil Rhodes]

> Make sure the source code corresponds exactly to the library    binaries you 
>are distributing.
How is this to be proven?
P
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Re: [FFmpeg-user] ERROR: libx264 not found

[Reindl Harald]

Am 26.01.2017 um 11:44 schrieb Vlad Rahmileviz:

Hello,

I'm trying to build FFMPEG with "--enable-libx264" option.
It fails on configure
($ ./configure --enable-asm --enable-yasm --enable-shared --disable-static 
--prefix=/c/ffmpeg --toolchain=msvc --arch=x86_64 --extra-cflags="-MD" 
--enable-gpl --enable-libx264)


pff - what about install x264-devel/x264-dev or however that package is 
calledn on your operating system?


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[FFmpeg-user] Report non Compliance with License

[tracy]

I'm attempting to contact someone with whom I can report a non
compliance with ffmpsg's License terms.

The following Company

VideoSurgeon at http://videosurgeon.net/jv/4/ (see attachment) appears
not to be in compliance with the terms of the License. The following
conditions do not appear to be met


Distribute the source code of FFmpeg, no matter if you modified it
or not.
Make sure the source code corresponds exactly to the library
binaries you are distributing.


Explain how you compiled FFmpeg, for example the configure line, in
a text file added to the root directory of the source code.
Use tarball or a zip file for distributing the source code.
Host the FFmpeg source code on the same webserver as the binary you
are distributing.
Add "This software uses code of http://ffmpeg.org>FFmpeg licensed under the http://www.gnu.org/licenses/old-licenses/lgpl-2.1.html>LGPLv2.1 and 
its source can be downloaded here" to every 
page in your website where there is a download link to your application.
Mention "This software uses libraries from the FFmpeg project under
the LGPLv2.1" in your program "about box".
Mention in your EULA that your program uses FFmpeg under the
LGPLv2.1.
If your EULA claims ownership over the code, you have to explicitly
mention that you do not own FFmpeg, and where the relevant owners can be 
found.
Remove any prohibition of reverse engineering from your EULA.
Apply the same changes to all translations of your EULA.
Do not misspell FFmpeg (two capitals F and lowercase "mpeg").
Do not rename FFmpeg dlls to some obfuscated name, but adding a
suffix or prefix is fine (renaming "avcodec.dll" to "MyProgDec.dll" is not 
fine, but to "avcodec-MyProg.dll" is).
Go through all the items again for any LGPL external library you
compiled into FFmpeg (for example LAME).
Make sure your program is not using any GPL libraries (notably
libx264).



regards

Tracy Barlow









Sent from my Ubuntu Phone
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Re: [FFmpeg-user] Static ARMEL ffmpeg build on Zero not resolving domains properly?

[christina zou]

Also want to add the DNS error that forces me to use IP addresses instead
of domains. Hopefully people know what I am referring to; a friendly helper
from the IRC channel was able to reproduce it himself.

*raspivid -o - -t 0 -vf -hf -fps 10 -g 30 -b 450 | ffmpeg/./ffmpeg -f
lavfi -re -i anullsrc -f h264 -thread_queue_size 1024 -framerate 10
-probesize 100  -i - -vcodec copy -acodec aac -g 20  -f flv
"rtmp://rtmp-api.facebook.com:80/rtmp/1202172759851050?ds=1_l=1=ATig_KiAVVHvPWse
"*

ffmpeg version 3.1.4-static http://johnvansickle.com/ffmpeg/  Copyright (c)
2000-2016 the FFmpeg developers

  built with gcc 5.4.1 (Debian 5.4.1-2) 20160904

  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --enable-libmp3lame --enable-libx264 --enable-libwebp
--enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype
--enable-fontconfig --enable-libxvid --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libtheora --enable-libvo-amrwbenc
--enable-gray --enable-libopus --enable-libass --enable-gnutls
--enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi
--disable-indev=sndio --disable-outdev=sndio --enable-librtmp --cc=gcc-5
--disable-ffplay

  libavutil  55. 28.100 / 55. 28.100

  libavcodec 57. 48.101 / 57. 48.101

  libavformat57. 41.100 / 57. 41.100

  libavdevice57.  0.101 / 57.  0.101

  libavfilter 6. 47.100 /  6. 47.100

  libswscale  4.  1.100 /  4.  1.100

  libswresample   2.  1.100 /  2.  1.100

  libpostproc54.  0.100 / 54.  0.100

Input #0, lavfi, from 'anullsrc':

  Duration: N/A, start: 0.00, bitrate: 705 kb/s

Stream #0:0: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s

[h264 @ 0x3d54900] Stream #0: not enough frames to estimate rate; consider
increasing probesize

[h264 @ 0x3d54900] decoding for stream 0 failed

Input #1, h264, from 'pipe:':

  Duration: N/A, bitrate: N/A

Stream #1:0: Video: h264 (High), yuv420p, 1920x1080, 10 tbr, 1200k tbn,
20 tbc

*Problem accessing the DNS. (addr: rtmp-api.facebook.com
)*

*rtmp://rtmp-api.facebook.com:80/rtmp/1202172759851050?ds=1_l=1=ATig_KiAVVHvPWse
:
Unknown error occurred*

On Thu, Jan 26, 2017 at 2:18 AM, christina zou 
wrote:

> Just want to add my command and the -report log.
>
> *Command*
>
> raspivid -o - -t 0 -vf -hf -fps 30 -b 600 | ffmpeg/./ffmpeg -re -ar
> 44100 -ac 2 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero -f h264 -i -
> -vcodec copy -acodec aac -ab 128k -g 50 -strict experimental -f flv
> rtmp://31.13.66.23:80/rtmp/1202172759851050?ds=1_l=1=ATig_KiAVVHvPWse
>
> *Log (report)*
>
> ffmpeg started on 2017-01-26 at 07:07:25
>
> Report written to "ffmpeg-20170126-070725.log"
>
> Command line:
>
> ffmpeg/./ffmpeg -report -re -ar 44100 -ac 2 -acodec pcm_s16le -f s16le -ac
> 2 -i /dev/zero -f h264 -i - -vcodec copy -acodec aac -ab 128k -g 50 -strict
> experimental -f flv "rtmp://157.240.3.12:80/rtmp/1202172759851050?ds=1"
>
> ffmpeg version 3.1.4-static http://johnvansickle.com/ffmpeg/  Copyright
> (c) 2000-2016 the FFmpeg developers
>
>   built with gcc 5.4.1 (Debian 5.4.1-2) 20160904
>
>   configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --enable-libmp3lame --enable-libx264 --enable-libwebp
> --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype
> --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libtheora --enable-libvo-amrwbenc
> --enable-gray --enable-libopus --enable-libass --enable-gnutls
> --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi
> --disable-indev=sndio --disable-outdev=sndio --enable-librtmp --cc=gcc-5
> --disable-ffplay
>
>   libavutil  55. 28.100 / 55. 28.100
>
>   libavcodec 57. 48.101 / 57. 48.101
>
>   libavformat57. 41.100 / 57. 41.100
>
>   libavdevice57.  0.101 / 57.  0.101
>
>   libavfilter 6. 47.100 /  6. 47.100
>
>   libswscale  4.  1.100 /  4.  1.100
>
>   libswresample   2.  1.100 /  2.  1.100
>
>   libpostproc54.  0.100 / 54.  0.100
>
> Splitting the commandline.
>
> Reading option '-report' ... matched as option 'report' (generate a
> report) with argument '1'.
>
> Reading option '-re' ... matched as option 're' (read input at native
> frame rate) with argument '1'.
>
> Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
> (in Hz)) with argument '44100'.
>
> Reading option '-ac' ... matched as option 'ac' (set number of audio
> channels) with argument '2'.
>
> Reading option '-acodec' ... matched as option 'acodec' (force audio codec
> ('copy' to copy stream)) with argument 'pcm_s16le'.
>
> Reading option '-f' ... matched as option 'f' (force format) with argument
> 's16le'.
>
> Reading option '-ac' ... matched as 

[FFmpeg-user] ffmpeg h264_vaapi hangs

[Nalorokk]

When I'm trying to encode any h264 video to h264 using h264_vaapi ffmpeg
hangs just after start. Is there any way to make hardware encoding working?


   1. ffmpeg -loglevel debug -hwaccel vaapi -vaapi_device
   /dev/dri/renderD128 -i output.mp4 -vf 'format=nv12,hwupload' -map 0:0
   -threads 8 -aspect 16:9 -y -f matroska -acodec copy -b:v 12500k -vcodec
   h264_vaapi out_va.mp4

   http://pastebin.com/2MmJBY04 - full log
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[FFmpeg-user] Static ARMEL ffmpeg build on Zero not resolving domains properly?

[christina zou]

Hello,

I have a Pi Zero and am using ffmpeg to stream to platforms like Youtube
and Twitch.

I was instructed from #ffmpeg's IRC channel to install the static ARMEL
ffmpeg tar from https://www.johnvansickle.com/ffmpeg/. It is called
ffmpeg-release-armel-32bit-static.tar.xz

- md5


Using this version of ffmpeg, I am able to livestream to platforms like
Youtube and Twitch Live, but I'm forced to use the IP addresses in the RTMP
URLs to avoid a DNS issue. This DNS issue was repro'd by others helping me
on the IRC channel, so they advised the IP address approach. They told me
the ARMHF version resolved this issue, but I would have to manually compile
it for the Zero, and it was complicated (my memory is hazy here...)

Now I am attempting to stream to Facebook Live, where it seems like the
RTMP URL's IP address changes, and I'm getting a 'failed to read RTMP
packet header' error.

I am trying to remember the names of the users I interacted with on
#ffmpeg.. I think I remember @furq?

I'm wondering if I should use a different version of ffmpeg, and if so, how
to obtain/build it for the Zero?

Thanks,
Christina
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[FFmpeg-user] Suggestions for FFmpeg: support a simpler command line for concatenating audio files with a specified overlap

[a]

Hello, after hunting around ffmepg.org, I could not find where to post 
suggestions for FFmpeg developers, so if this mailing list is the appropriate 
place here you go otherwise please let me know:


  1.  I have a need to concatenate programmatically several short (< 1 second) 
mp3 files but with a specific overlap for example 100ms, the current solution 
is a command line of this form
 *
1.  ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3
2. -filter_complex
3.   "[1]adelay=delay1[b];
4.[2]adelay=delay2[c];
5.[3]adelay=delay3[d];
6.[0][b][c][d]amix=4"
7.  out.mp3


 *   However this forces me to derive each file durations and do the math.  
I was hoping if you would consider supporting something that would not require 
retrieving the durations and just specify the relative delays during 
concatenation.  Here are a couple of examples

 *   Delay all input file by the same amount
ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3 -filter_complex "delay=100; amix=4"

 *   Delay each input by the specified amount or zero if not specified.
ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -i 4.mp3 -filter_complex "[1]delay=100; 
[2]delay=20; amix=4"

 *   Of course this may not be generalized enough to offer the most 
flexibility so I will let you decide on the final form of the options, but you 
get the point.

 *   NOTE: I am using C-Sharp to build the command line for other FFmpeg 
tasks, however the simpler command line above would allow easier prototyping 
from a batch file without requiring hardcoding exact value for delay1, 
delay2,, delayN
Thank you.
M.
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Re: [FFmpeg-user] Static ARMEL ffmpeg build on Zero not resolving domains properly?

[christina zou]

Just want to add my command and the -report log.

*Command*

raspivid -o - -t 0 -vf -hf -fps 30 -b 600 | ffmpeg/./ffmpeg -re -ar
44100 -ac 2 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero -f h264 -i -
-vcodec copy -acodec aac -ab 128k -g 50 -strict experimental -f flv
rtmp://31.13.66.23:80/rtmp/1202172759851050?ds=1_l=1=ATig_KiAVVHvPWse

*Log (report)*

ffmpeg started on 2017-01-26 at 07:07:25

Report written to "ffmpeg-20170126-070725.log"

Command line:

ffmpeg/./ffmpeg -report -re -ar 44100 -ac 2 -acodec pcm_s16le -f s16le -ac
2 -i /dev/zero -f h264 -i - -vcodec copy -acodec aac -ab 128k -g 50 -strict
experimental -f flv "rtmp://157.240.3.12:80/rtmp/1202172759851050?ds=1"

ffmpeg version 3.1.4-static http://johnvansickle.com/ffmpeg/  Copyright (c)
2000-2016 the FFmpeg developers

  built with gcc 5.4.1 (Debian 5.4.1-2) 20160904

  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --enable-libmp3lame --enable-libx264 --enable-libwebp
--enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype
--enable-fontconfig --enable-libxvid --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libtheora --enable-libvo-amrwbenc
--enable-gray --enable-libopus --enable-libass --enable-gnutls
--enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi
--disable-indev=sndio --disable-outdev=sndio --enable-librtmp --cc=gcc-5
--disable-ffplay

  libavutil  55. 28.100 / 55. 28.100

  libavcodec 57. 48.101 / 57. 48.101

  libavformat57. 41.100 / 57. 41.100

  libavdevice57.  0.101 / 57.  0.101

  libavfilter 6. 47.100 /  6. 47.100

  libswscale  4.  1.100 /  4.  1.100

  libswresample   2.  1.100 /  2.  1.100

  libpostproc54.  0.100 / 54.  0.100

Splitting the commandline.

Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.

Reading option '-re' ... matched as option 're' (read input at native frame
rate) with argument '1'.

Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '44100'.

Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.

Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'pcm_s16le'.

Reading option '-f' ... matched as option 'f' (force format) with argument
's16le'.

Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.

Reading option '-i' ... matched as input file with argument '/dev/zero'.

Reading option '-f' ... matched as option 'f' (force format) with argument
'h264'.

Reading option '-i' ... matched as input file with argument '-'.

Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.

Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'aac'.


  libswresample   2.  1.100 /  2.  1.100

  libpostproc54.  0.100 / 54.  0.100

Splitting the commandline.

Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.

Reading option '-re' ... matched as option 're' (read input at native frame
rate) with argument '1'.

Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '44100'.

Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.

Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'pcm_s16le'.

Reading option '-f' ... matched as option 'f' (force format) with argument
's16le'.

Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.

Reading option '-i' ... matched as input file with argument '/dev/zero'.

Reading option '-f' ... matched as option 'f' (force format) with argument
'h264'.

Reading option '-i' ... matched as input file with argument '-'.

Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.

Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'aac'.

Reading option '-ab' ... matched as option 'ab' (audio bitrate (please use
-b:a)) with argument '128k'.

Reading option '-g' ... matched as AVOption 'g' with argument '50'.

Reading option '-strict' ...Routing option strict to both codec and muxer
layer

 matched as AVOption 'strict' with argument 'experimental'.

Reading option '-f' ... matched as option 'f' (force format) with argument
'flv'.

Reading option 'rtmp://157.240.3.12:80/rtmp/1202172759851050?ds=1' ...
matched as output file.

Finished splitting the commandline.

Parsing a group of options: global .

Applying option report (generate a report) with argument 1.

Successfully parsed a group of options.

Parsing a group of options: input file /dev/zero.

Applying 

Re: [FFmpeg-user] Convert MPEG-PS progressive to Interlaced

[Rens Dijkshoorn]

> Hi All,
> I have been struggling with converting PS progressive stream to PS
> interlaced for my test app.
> 
> But all the conversion commands used by me are converting to Frame based
> interlaced stream but what I am looking for is I-P field based interlace(I
> frame having top filed and next frame having the bottom field) .

Maybe we are talking about different things

Interlaced = 2 fields one top one bottom together is 1 frame.
Changing a PS stream to interlaced would create a PSF frame
The 2 fields that make one frame don't have any temporal difference


ffmpeg has a few filter commands in this direction
have a look at a combination of framerate and tinterlace

-vf framerate=50,tinterlace=4

I-frames or P-frames has more to do with mpeg encoding that
with the video being interlace.

For proper interlaced encoding the encoder needs to be
interlaced aware of this field structure and the possiblity
of temporal difference between fields in one video frame. 

Regards

Rens
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