Hello everybody, as I've said before, I am a student and I'm triyng to
learn how to use ffmpeg.
It is not an easy task, because there is not a lot of documents that
explain it in a easy way.
I would like to share with you the tools I found useful, and I ask you to
reply linking the resourcies and t
; On Feb 18, 2021, at 03:32, Marco Mircoli wrote:
> >
> > Hello,
> > I was just asking an help, something like...a weblink to something
> > similar I could adapt, an example.
> > I am a student, I am learning by myself. It's not easy, because online
> > ther
o gio 18 feb 2021 alle ore 12:02 Paul B Mahol ha
scritto:
> On Thu, Feb 18, 2021 at 11:33 AM Marco Mircoli
> wrote:
>
> > Hello,
> >tried a lot of times, but nothingI can't do it well.
> > Any advice?
> > The goal is to do it with a single comman
Hello everibody,
tried few times to use it, never succeed :-(
Read on internet all the document available on this filet, but nothing.
Any help?
Thanks.
S.
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Hello,
tried a lot of times, but nothingI can't do it well.
Any advice?
The goal is to do it with a single command line.
Thanks.
S.
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I think it is not a bug.
It is normal that dynaudnorm and loudnorm gives differents results, they
are 2 different instruments and they have different parameters and actions.
Thanks.
S.
Il giorno dom 14 feb 2021 alle ore 12:08 Paul B Mahol ha
scritto:
> On Sun, Feb 14, 2021 at 12:01 PM Ma
Yes, the result Is not leveled to the target loudness value
Il Ven 12 Feb 2021, 19:26 Paul B Mahol ha scritto:
> On Fri, Feb 12, 2021 at 7:17 PM Marco Mircoli
> wrote:
>
> > Il giorno ven 12 feb 2021 alle ore 10:53 Paul B Mahol
> > ha
> > scritto:
> >
>
Il giorno ven 12 feb 2021 alle ore 10:53 Paul B Mahol ha
scritto:
> On Fri, Feb 12, 2021 at 12:48 AM Marco Mircoli
> wrote:
>
> > Hello everybody,
> > wondering how to set the same output audio specs (sample rate anche bit
> > depth) as input.
> >
> &
Hello everybody,
wondering how to set the same output audio specs (sample rate anche bit
depth) as input.
I tried like that
ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
5:TP=-1.5:LRA=7 -sample_fmt s16 -ar 44100 FILE_TO
if I have a 48KHZ input, the output is 44100, so doesn't work for my
Hello,
wandering if it is possible to process just the 1st audio channel of an
input.
I don't know how many channels I will have as input, but I know that I have
to process just the 1st and the others I have not to considerate.
INPUT from mono to multichannel
filters.
OUTPUT just one channe
t it does not
> have that specific filter.
> Also note that those audio filters are not magic way to solve your problems
> by putting them in filtergraph and forgetting about them next second.
>
>
> > On Fri, Jan 29, 2021, 10:21 PM Paul B Mahol wrote:
> >
> > >
Mahol ha
scritto:
> On Fri, Jan 29, 2021 at 11:25 AM Marco Mircoli
> wrote:
>
> > Hello,
> > anybody knows the reason why this command works.
> >
> > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> >
> >
> adeclick,afftdn=nr=80:nf
.
Il giorno gio 28 gen 2021 alle ore 10:15 Michael Koch <
astroelectro...@t-online.de> ha scritto:
> Am 28.01.2021 um 09:44 schrieb Marco Mircoli:
> > Goodmorning everyone.
> > I'm new to ffmpeg and was wondering if anyone from the group could help
> me
> > set up
or some fairly complex situations and, in the logs, provides the
> ffmpeg command that it used to get there.
> My need at the time was video specific. I'm unsure if it supports all the
> audio filters you're looking for.
>
> -Jason
>
>
>
> On Thu, Jan 28,
ng?
Thanks.
S.
Il giorno gio 28 gen 2021 alle ore 10:24 Chris Miceli
ha scritto:
> Hi Marco,
>
> If you are a user of MacOS then have a look at Handbrake. I think there is
> a windows version as well but as a Linux user I'm not 100% sure.
>
> If you hunt for "handbr
Hello,
my problem is that I am not a programmer (I'm a sound designer)
I would like to approach FFMPEG.
It's hard for me to learn the syntax of ffmpeg.
Do you know if is there a tool that let me use ffmpeg filters in a frontend
environment and shows me the command line of what I'm doing?
Thanks
Goodmorning everyone.
I'm new to ffmpeg and was wondering if anyone from the group could help me
set up a command line.
My goal is to improve voice audio recordings.
I would like to be able to do this starting from a command line I have in
my script, modifying it to add:
- 70HZ high pass filter
-
Good morning
Who can I write to for help?
M.
*--Marco Sestan*
Director
*Audio-Video Production& PostProduction Manager*
*Mob: +39 348 47 39 003*
Il giorno sab 24 ott 2020 alle ore 01:45 Moritz Barsnick
ha scritto:
> Hi Daniel,
>
> On Wed, Oct 21, 202
As plugin I mean LADSPA plugin
Il giorno sab 3 ott 2020 alle ore 13:46 Marco Mircoli
ha scritto:
> Hello,
>quite interested if there is a de-reverb funcion in ffmpeg or if is it
> possible to implement via plugins.
> Thank
Hello,
quite interested if there is a de-reverb funcion in ffmpeg or if is it
possible to implement via plugins.
Thanks in advance.
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etadata
Publisher='agency' -metadata Language=italian -b:a ".$convert_speed."k
$audio_output_mp3 2>&1");
Now, I would like to insert an album artwork in the mp3.
Is it possible?
Thanks.
S.
Il giorno gio 1 ott 2020 alle ore 17:45 Marco Mircoli
ha scritto:
> Hello, t
udio_output_mp3 2>&1");
Tried with 700 hz
the result is very low in volume
it seems that it normalize to -24 and after apply the highpass
another point I can't fix is the metadata, they works except for the
author:url.
Maybe am I wrong with syntax?
Thanks in advance.
Il giorn
Thanks a lot!
Il giorno gio 1 ott 2020 alle ore 14:17 Michael Koch <
astroelectro...@t-online.de> ha scritto:
> Am 01.10.2020 um 14:06 schrieb Michael Koch:
> > Am 01.10.2020 um 13:25 schrieb Marco Mircoli:
> >> Hello',
> >> I'm a newbie.
&g
Hello',
I'm a newbie.
Just bought a php script that use ffmpeg.
it converts to mp3/96Kbps all media uploaded in a unique format.
this is the line, and now it normalizes to R128 (thanks Moritz)
$shell = shell_exec("$ffmpeg_b -i $audio_file_full_path -map 0:a:0 -af
loudnorm -b:a 96k $audio_o
Thank you!
Il giorno gio 24 set 2020 alle ore 09:54 Paul B Mahol ha
scritto:
> On Wed, Sep 23, 2020 at 03:30:25PM +0200, Marco Mircoli wrote:
> > Thanks Motitz,
> >this is a one pass norm.
> > Is there a way to implement 2 pass?
>
> Yes there is a way, you ca
Thanks Motitz,
this is a one pass norm.
Is there a way to implement 2 pass?
Thanks.
S.
Il giorno mer 23 set 2020 alle ore 14:42 Moritz Barsnick
ha scritto:
> On Wed, Sep 23, 2020 at 14:16:31 +0200, Marco Mircoli wrote:
> > Just bought a php script that use ffmpeg.
>
> I hop
Hello',
I'm a newbie.
Just bought a php script that use ffmpeg.
it converts to mp3/96 all media uploaded in a unique format.
this is the line
$shell = shell_exec("$ffmpeg_b -i $audio_file_full_path -map 0:a:0 -b:a
96k $audio_output_mp3 2>&1");
I'm wondering if it is possible to include in
,agroup:audio”
$SEGMENT_FILE_NAME
$MEDIA_PLAYLIST_PREFIX
Please help me to fix this behavior.
Best regards
Marco Kittel
pEpkey.asc
Description: application/pgp-keys
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options, in order to say in the last where via var_map how
my content is arranged in a media playlist. Is their a way to that
ffmpeg implicitly takes the video track and takes all audio content and
maps it to hls?
Best regards
Marco Kittel
pEpkey.asc
Description: application/pgp-keys
. the whole 250 GOP is written at
once. Is there any way to force buffers to be flushed more frequently without
increasing the number of i-frames? I would like to start processing the data as
soon as possible rather than having to wait for all 250 frames to be processed.
- Marco
idering my target to place the four video together, do you have any idea to
speed up the file creation?
Regards
On Sunday, November 26, 2017, 12:49:17 PM GMT+1, Carl Eugen Hoyos
wrote:
2017-11-26 8:51 GMT+01:00 Marco De Angelis :
> I'm analyzing the video using Quicktime since
Well, thank you! FFmpeg is an impressive project.
-- Marco
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On Mon, Nov 27, 2017 at 2:43 AM, Carl Eugen Hoyos wrote:
> Please double the definition of
> ALSA_BUFFER_SIZE_MAX
> on top of libavdevice/alsa.h until the device works for you
> and report back so I can commit the new value!
Doubling once from 65536 to 131072 was enou
million
frames, which is way more than what is really needed and causes a memory
allocation error on my system.
:04 04 bc8ae5608ccfe6b46dc90633c535d91b4c6c0e2b
88c1baef83722b3310430a9933221876d0db6095 M libavdevice
Best,
Marco
On Sun, Nov 26, 2017 at 8:02 PM, Carl Eugen Hoyos wrote:
>
> Then please use git bisect to find the change
> introducing the regression.
I'm trying, but many commits do not compile properly.
Specifically, I'm stuck with:
make: *** No rule to make target 'libavutil/macros.h', needed by
'libswscal
>
> > Thanks! With this, 0.6.7 runs
>
> Sorry, I don't understand this sentence.
Using the option '-ac 2', ffmpeg 0.6.7 works.
But, using the same option, the current version from the master git branch
doesn't work.
-- Marco
> $ ffmpeg -f alsa -ac 2 -i hw:1,0 -t 30 out.wav
Thanks! With this, 0.6.7 runs but newer versions still complain about:
[alsa @ 0x55eb8e2a6780] cannot set ALSA buffer size (Invalid argument)
hw:1,0: Input/output error
Any idea?
-- Marco
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ffm
0
configuration: --disable-vaapi --disable-ffserver
--prefix=/home/marco/ffmpeg_build
--extra-cflags=-I/home/marco/ffmpeg_build/include
--bindir=/home/marco/bin
libavutil 50.15. 1 / 50.15. 1
libavcodec52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 5
: --prefix=/home/marco/ffmpeg_build
--pkg-config-flags=--static
--extra-cflags=-I/home/marco/ffmpeg_build/include
--extra-ldflags=-L/home/marco/ffmpeg_build/
lib --extra-libs='-lpthread -lm' --bindir=/home/marco/bin --enable-gpl
--enable-libass --enable-libfdk-aac --enable-libfreetyp
: subdevice #0
card 1: Em28xxAudio [Em28xx Audio], device 0: Em28xx Audio [Empia 28xx Capture]
Subdevices: 1/1
Subdevice #0: subdevice #0
and I can capture audio/video without problems using the `qv4l2` test bench.
Thanks!
Marco
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30991.07[aac @ 01df09e10740] Qavg: 1909.973
On Thursday, November 23, 2017, 1:24:27 PM GMT+1, Carl Eugen Hoyos
wrote:
2017-11-23 12:52 GMT+01:00 Marco De Angelis :
> C:\S7\ffmpeg\bin\ffmpeg -i C:\S7\FTP_video\1.mp4 -i C:\S7\FTP_video\2.mp4
> -i C:\S7\FTP_video\3.mp4 -i C:\S7\
This is the command
C:\S7\ffmpeg\bin\ffmpeg -i C:\S7\FTP_video\1.mp4 -i C:\S7\FTP_video\2.mp4 -i
C:\S7\FTP_video\3.mp4 -i C:\S7\FTP_video\4.mp4 -filter_complex
"nullsrc=size=1920x1080 [base]; [0:v] setpts=PTS-STARTPTS, scale=960x540
[upperleft]; [1:v] setpts=PTS-STARTPTS, scale=960x540 [upperrig
Hello, I'm pretty new to video processing and I came across this usuful tool.
Using the info I found on the following page I created a mosaic using four mp4
files (original frame rate is 60 fps).
Create a mosaic out of several input videos – FFmpeg
|
|
|
| | |
|
|
|
| |
Create a mosa
Found the problem. SOLVED.
Ignore last thread, was a misconfiguration.
Il giorno ven, 14/04/2017 alle 02.56 +0200, Marco ha scritto:
> Hello,
>
> i updated my Ubuntu to 17.04
>
> I have a self-compiled FFMPEG with this script:
> https://pastebin.com/hQekxBsN
>
> I hav
advance. Sorry for bad english
Marco.
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Hello,
i use to encode the audio of my movie using libfdk_aac, but in the
result file doesn't
appear the used bitrate (with mediainfo, ffprobe, etc.). I encode using
CBR mode, in general
matroska container.
I have the same problem with Handbrake.
I've written to the FDK site, this is the respons
bad english. Thanks for your attention.
--
Marco Diletti
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actually an issue related to the
configure script.
> Note that since compilation does not succeed for all files, I
> suggest you use gcc for the time being.
I manually patched the configure script to bypass "check_as" and I also
realized the compilation fails. That's the
o
/var/folders/zn/45tzvzd15tnfvp4g5847x2dmgn/Tffconf.Ct6YZsBn.o
/var/folders/zn/45tzvzd15tnfvp4g5847x2dmgn/Tffconf.a7i3f9zc.S
/var/folders/zn/45tzvzd15tnfvp4g5847x2dmgn/Tffconf.a7i3f9zc.S:1:1:
error: unknown directive
.altmacro
GNU assembler not found, install/update gas-preprocessor
The weird thing is that *if I create a file containing .altmacro and run
the same exact command it actually works*. For example:
$ echo ".altmacro" > test.as
$ /tmp/toolchain/bin/arm-linux-androideabi-clang
--sysroot=/tmp/toolchain/sysroot -isysroot /tmp/toolchain/sysroot
-D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
-Dstrtod=avpriv_strtod -DPIC -I/sysroot/usr/include -ffast-math
-funroll-loops -march=armv7-a -mfloat-abi=softfp -mfpu=vfpv3-d16
-march=armv7-a -fPIC -c -o /tmp/test.o test.as
$ echo $?
0
After digging for an hour in the configure script I can't actually realize
why it fails. Do you have any hint?
Thank you,
Marco
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sp "rtmp://localhost:1935/live/132646"*
All works well!!!
Why if I specify the rtsp_transport works well and why in the first case
doesn't work?
Thanks,
Marco
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rawvideo -c:v rawvideo -pix_fmt gray16le -y
out16_decoded.yuv
the resulting files are not identical.
Any explanation for this behavior? Am I doing something wrong?
Best regards,
--Marco
D:\Users\Porsch\Develop\LoggingApp\tools\build\vs12\dl4log_mux>d:\Tools\ffmpeg-20150109-git-d1c6b7b-wi
On 20/08/14 17:59, Marco Baumgartl wrote:
The video stream (duration=3176.106267) is shorter than the audio stream
(duration=3221.490068). While the source video is in sync, the
synchronization is lost during concatenation.
I "solved" the problem by adjusting stream durations using
Am 20.08.2014 05:49, schrieb Qianliang Zhang:
You must scale to the same size before you concat
Yes, I know. The videos have the same resolution before concatenation.
Marco
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11:44
TAG:language=und
[/STREAM]
Cheers,
Marco
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,
Marco
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Hi!
I need to concat screencast recordings with an intro and an outro video.
Unfortunately the final video doesn't play nicely because the outro is
visible before the original video is finished (while the audio of the
screencast recording continues).
These are my commandline options:
ffmpeg
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