ok, thanks!
Em qui., 2 de mai. de 2024 às 00:21, Carl Zwanzig escreveu:
> On 5/1/2024 7:55 PM, CMG DiGiTaL wrote:
> > How do I generate the log, but keep the conversion information on the
> > screen as well?
>
> (really a windoze question,
I run a command to normalize an Audio file and write the output to a log
file according to the level I want.
Full command:
ffmpeg -hide_banner -i "C:\Users\%username%\Desktop\Music.mp3" -af
"loudnorm=linear=true:I=-10.0:LRA=11:tp=-0.1:measured_I=-9.8:measured_LRA=2.8:measured_tp=+0.5:measured_thr
>
> You said:
> *-report file=test.log:level=16* the character * on the commandline
> causes ffmpeg to look at this as a outputfile specification to put
> decoded results in, as nothing is specified about how to decode it it
> tries to guess what codecs to use on the extension. Naturally this does
there is no * in my code, this appears because I put the command in bold to
highlight it!... how do I escape the colon?
Em sáb., 24 de fev. de 2024 às 17:25, Carl Zwanzig escreveu:
> On 2/24/2024 9:16 AM, CMG DiGiTaL wrote:
> > [NULL @ 02738af86200] Unable to find a suitable outp
OK,
but how to make the names of the log files, which will be generated in the
Normalizando_lufs folder, match the names of the mp3 audios, e.g.:
Blalala.log
Cracrcra.log
Dedede.log
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Em sáb., 24 de fev. de 2024 às 17:25, Carl Zwanzig escreveu:
> Always start by posting the complete ffmpeg command and unedited output,
> that way we don't have to guess about what the script is doing.
>
> z!
> ___
>
> My bat file will read several mp3
hi,
I created a bat file where I enter the LUFS values to cover several audio
files in a folder.
However, I would like to generate an error log file.
I used the command -report file=log.txt:level=16.
see below the command in my bat file:
md "C:\Users\%username%\Desktop\Normalizing_lufs"
pushd "%
Hi,
I'm using ffmpeg normalize loundnorm and some audios show "innacurete" or
"1152 samples" warning messages.
What I wanted to know is if there is a command that I can use to generate a
"log.txt" of only the files with alert messages, for my control?
Thank's
Claudio.
Hi Paul,
But it's not me who defines which mode will be used, but Loudnorm... how
can I do that then?
Using dynamic mode and not linear is far from working correctly.
>
> One use dynamic mode when not specifying measured values to filter.
>
Hi Paul,
__
I searched the internet before and couldn't find any comments about this
situation:
In some songs, the loudnorm filter quickly increases the volume of part of
the audio and returns to normal in milliseconds.
Has anyone been in this situation or know what could be happening?
__
> First you top post, second you call names.
>
> I wanted to help you, but I see only powerless person than need help and
> knowledge.
>
Yeah, and you're a badass on the ffmpeg list, congratulations to you. You
talk like a mule!
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Bouke / Videotoolshed <
bo...@videotoolshed.com> escreveu:
> Learn to live with it, as well as the flaming for top posting, as I’m
> about to receive.
> (Or not, if I tell I actually like it to be spanked by nerds.(
>
> Bouke
>
>
> > On 12 Apr 2022, at 19:02, CMG DiGiTaL wrote:
What is the real difference between the Loudnorm and EBU LUFS normalization
filters?
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ffmpeg-user-requ...@ffmpeg
I have noticed, from the harsh and uneducated answers, that the
distinguished
Mr. Paul B Mahol, must have serious social relationship problems...
...poor man, must be a very lonely person!
Em ter., 12 de abr. de 2022 às 07:54, Reindl Harald
escreveu:
>
>
> Am 12.04.22 um 12:29 schrieb Paul B M
I've been doing tests with LUFS normalizations using ebu r128 and loudnorm.
I still haven't seen any significant difference in relation to the
variation in dB loudness.
The command I use for normalization with ebu is:
ffmpeg-normalize "input.mp3" -nt ebu -t -10.0 -tp -1.0 "output.mp3"
Does ffmpeg
‘-y (*global*)’
>
> Overwrite output files without asking.
>
hi François,
This command I was already using. It turns out that we have two situations:
the message from the ffmpeg command
itself and the problem of the explorer creating the duplicate output files
with different names.
The "-y" ha
Em qui., 7 de abr. de 2022 às 10:55, CMG DiGiTaL
escreveu:
> I use the command below to generate the LUFS normalization file. Note that
> the output file will be generated in the same folder as the renamed input
> file.
> Command:
>
> FOR /F "tokens=1,2 delims=,
I use the command below to generate the LUFS normalization file. Note that
the output file will be generated in the same folder as the renamed input
file.
Command:
FOR /F "tokens=1,2 delims=," %%b IN ('ffprobe -v 0 -select_streams a
-show_entries "stream=sample_fmt,sample_rate" -of "csv=p=0"
"!fil
Ok Paul, nice to know about the FLAC encoder, I'll check it out!
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>
> well, that's a number for new tracks where i consider before or after
> the first coffee - OK, to be honest for 1800 songs it may take two hours
> including start playing and flag ballads in the ID3-Tags :-)
>
> 83490 tracks
> 78796 unique tracks (filtered live/unpligged..)
> 76069 lyrics
>
> 2
Em qua., 30 de mar. de 2022 às 07:27, CMG DiGiTaL
escreveu:
> Better use ebur128 scanner filter, it is much faster, after it reports
>> values, just apply volume filter.
>>
>
>
>> loudnorm dynamic processing is not perfect.
>
>
> hi Paul,
> Really?... base
>
> I would rather use FLACs original frontend (than ffmpeg), which also
> provides a nice Windows GUI interface. FLAC conversion is also included
> in a variety of software products, i.e. Exact Audio Copy, which I used
> to rip all my music CDs.
>
Hi Wolfgang,
I also use Exact Audio Copy, the go
> Better use ebur128 scanner filter, it is much faster, after it reports
> values, just apply volume filter.
>
> loudnorm dynamic processing is not perfect.
>
hi Paul,
Really?... based on the command line I sent, how could I do it the way you
are saying?... can I use the same
command changing s
> loudnorm changes the sample rate of the output file to 192000 Hz. I use
> the tokens to select the value of the sample rate of the input file to
> generate the output file correctly, however, the command is giving an error.
>
Informing everyone that I found the error. The correct syntax for the
>
> no - what you need to understand is what you deal with, no matter the
> topic - a foll with a tool is still a fool :-)
>
> in case of a LOSSLESS codec the bitrate don't matter - that's common
> sense - translate it to compression efficiency and by common sense it
> depends on the content
Ok
loudnorm changes the sample rate of the output file to 192000 Hz. I use the
tokens to select the value of the sample rate of the input file to generate
the output file correctly, however, the command is giving an error.
command:
FOR /F "tokens=1,2 delims=," %%b IN ('ffprobe -v 0 -select_streams a
>
> seems like you still didn't understand the repsonse to your other thread
> - FLAC is *lossless* - forget your MP3 knowledge when you deal with FLAC
>
Reindl Harald,
What I really need to understand is Mediainfo!
thanks
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>
> Default compression value in ffmpeg is 5
> So for maximum lossless compression use
> ffmpeg -i "input.wav" -af aformat=s16:44100 -c:a flac -compression_level
> 12 "output.flac"
>
Okay Ferdi, thanks.
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I'm using the command below to convert from wav to flac:
ffmpeg -i "input.wav" -af aformat=s16:44100 "output.flac"
The bit-depth and sample_rate are ok, but is the command converting the
audio from CBR to VBR.
What should I add in the above command, or what command should I use for
the audio to k
>
> As Flac is a lossless compression algorithm it has no fixed bitrate, the
> bitrate will change dependant on the content and cannot be set in ffmpeg.
> Best, Stuart
>
OK, ROBINSON Stuart.
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Em ter., 29 de mar. de 2022 às 05:16, CMG DiGiTaL
escreveu:
> I made a batch for LUFS normalization that works perfectly for mp3 and I'm
> creating another one to normalize flac files. The normalization of flac
> files also works perfectly, but in the output files, the flac codec
I made a batch for LUFS normalization that works perfectly for mp3 and I'm
creating another one to normalize flac files. The normalization of flac
files also works perfectly, but in the output files, the flac codec changes
the bit-depth to 24 and the bitrate also changes.
I use tokens to retrieve
> Not an ffmpeg question, but...
> Nope, it doesn't. I think you could make the set '(*.mp3 *mp4 *.wav)'
> etc...
>
OK thanks
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hi,
I have a bat that converts LUFS from audio files, but it only converts to
.mp3 files. How do I get it to convert any type of audio?
I tried to add two variables in the for command, but it doesn't work,
eg.
In:
FOR %%a IN (*.mp3) DO (
SET "filename=%%~na"
I tried:
FOR %%a %%e IN (*.*) DO (
>
> You could probably use vorbiscomment. I have no experience with the tool
> however.
>
OK, I'm already checking the vorbiscomment, and from my tests, it looks
like it will work.
I'm setting up a script that will read an image and prepare it with the
metada_block_picture
specs, so that vorbisc
>
> Now I understand why you posted 2 logs..
Exactly!... it was to show that the batch, when the file had a cover art,
read the same record twice.
.And do you really need -vsync vfr ? I'm no expert, but I think it's
> totally useless in this situation.
>
Now you don't need it anymore, I alread
Em qui., 10 de mar. de 2022 às 20:52, Reindl Harald
escreveu:
>
> volume normalization is easy to understand:
>
>
hi Reindl,
I'm doing a critical inspection here and testing the audio capture
microphones to see why these differences in some
audios. The ones I work with, until they are few, the p
>
> it adjusts the *general volume* of a track
> it's thes ame as touch the volume vontrol
> typically it *lowers* the volume of every file
>
ok Reindl,
It could be that these files were already peaking high and I just realized
now that I'm
paying more attention to my audio volumes.
thanks
>
>
> Yeah, this place seems kind of strange. Attacking people is ok, but don't
> you dare top post! Heh.
>
I fully agree with you!
>
> I understand what Clayton said about the volume on the amplifier, but
>
>
Thank you Clayton,
Got what I meant! ... as I have some very very low FLAC's,
Ok boys each one has its truth and the sum of them, makes us reach
excellence!... but in the end,
we must be faithful to what seems most correct to us.
Before starting this topic, I had done a lot of testing with software and
plugins related to LUFS normalization:
- The plugin I specified h
>
> Two decades ago I was heavily involved in a music restoration project of
> live concert performances
>
OK mark, very nice to hear about your audio experience!...I love it, it
makes me feel free and alive!
With this internet audio standardization movement, I'm paying more
attention to this typ
>
>
> You cannot increase the volume of a file that is already near/at full scale
>
ok Clayton,
I understood everything you said... I mentioned the plugin, because I used
it in some audio and it was really satisfactory,
as it is an evaluation product, that is, it will lose its functions in a
few
>
> Linear processing is multiplying whole file with single float number,
> volume/gain adjustment.
>
> Dynamic processing is changing dynamics of input audio.
>
hi Paul,
I understand what you said, but as I mentioned, I have some doubts related
to the test I did on the WLM Plus Stereo
plugin.
W
hi,
I'm trying to convert some mp3 to ogg add mp3 cover art on ogg, but I can't
do it. When I use the standard command, the mp3 art is transformed into
video on ogg and that's not what I want, see the command that generates the
mp3 art as a video on ogg:
ffmpeg -vsync 2 -i "input.mp3" -b:a 320k "
Yeah, this is what I suspected. You've got source files that are below
>
Okay Clayton, I get the concept!
As I'm doing a batch for myself, if necessary, normalize the LUFS of my
files, I was
left with this doubt as to why I can't do it in linear mode and why I
couldn't reach the -10 LUFS target.
>
> Post measured values of your flac files and target values you set?
>
I use the dual-pass normalization process, loudnorm analyzes the input
audio and generates
the values that will enter pass 2.
First pass:
An observation:
In this pass, I still haven't entered the LUFS target value or the Tr
> I haven't looked at this in ffmpeg, but in other audio circles whenever
>
OK,Clayton
the way you explained the behavior of loudnorm makes sense, it tries to
preserve the audio quality without cuts and compressions!
thanks
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hi,
About loudnorm normalization: Normalization in Dual-pass mode.
Rules for normalization to be linear:
1 - Values entry:
LUFS_TARGET (user entry)
LRA (get input file - Pass 1 loudnorm)
TP_TARGET (user entry)
THRESH (get input file - Pass 1 loudnorm)
OFFSET
nishing this program, the CONVERTED FILE LUFS
will be on your ^(Desktop^)
echo in the folder named "Converted Audio LUFS" so that you can
move it wherever you want on your PC.
echo After moving the file(s), you can delete the
Converted Audio LUFS folder.
echo.
echo.--
> I don't know why you're posting 2 logs.
Ok sorry, I posted both codes so I can verify that the code works.
> For the second one your Batch code doesn't appear to work correctly for
the bitrate and samplerate.
what is happening is that it seems that when the file has cover art, the
batch is doin
hi,
Problem solved!
I was able to get the image to be added to the MKV file without it being
created as a video stream. See command below:
ffmpeg -i "input.mkv" -attach "cover.jpg" -map 0 -c copy -metadata:s:t
mimetype="image/jpg" -metadata:s:t:0 filename="cover.jpg" "output_cover.mkv"
thanks
hi,
why when I add a cover art to an mkv file does ffmpeg convert the image
into a video stream? What I want is for the image to be attached to the
file as mjpeg only!
Command used:
ffmpeg -i "input_without_cover.mkv" -i "cover_image.jpg" -map 0:0 -map 1 -c
copy -id3v2_version 3 -metadata:s:v tit
>
> > [...]
> > -c:v copy -map 0:0 -acodec mp3 -b:a -ar:a N/A
> Apparently you did not.
>
>
sorry I sent the wrong log, follow the right log:
ffmpeg started on 2022-02-28 at 22:29:54
Report written to "ffmpeg-20220228-222954.log"
Log level: 48
Command line:
*ffmpeg *-report -i "Lob\xe3o - Cuidado
>
>
> With -map 0:0 you tell FFmpeg to process only the audio-stream. Remove it.
>
>
hi Reino,
I removed the -map 0:0 and the same message continues, below the message I
sent a log of the file:
[NULL @ 013de58f8b40] Unable to find a suitable output format for 'N/A'
N/A: Invalid argument
log:
>
>
> > Is there a way to check if the file has a cover and if so, record the
> cover
> > along with it?
> Simply add -c:v copy.
>
hi Reino,
I entered -c:v copy in command line as you suggested see below:
FOR /F "tokens=1,2 delims=," %%b IN ('ffprobe -v 0 -show_entries
stream^=bit_rate^,sample_ra
Is there a way to check if the file has a cover and if so, record the cover
along with it?
conversion command line:
FOR /F "tokens=1,2 delims=," %%b IN ('ffprobe -v 0 -show_entries
stream^=bit_rate^,sample_rate -of csv^=p^=0 "!filename!.mp3"') DO ffmpeg
-hide_banner -i "!filename!.mp3" -af
"loudno
> I'm glad it's now working for you.
>
ok Reino
and thanks for the help I got from all of you!
Clamarc
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ffm
hi Clayton,
I understand... rules are rules and if I've broken any, I apologize to
everyone!
thanks a lot for the help, the code has now read the file with the
exclamation mark no problems!
only when recording the file he was converting all to 48000kHz sample rate,
but I made a change in the re
t. On this mailinglist it's against the rules, to
> which you agreed upon subscribing.
>
> On 2022-02-22T07:32:05+0100, CMG DiGiTaL wrote:
> >> FOR *%G* IN (*.mp3) DO (
> >it was because I took part of the sample rate command that you sent me
> > to be able to us
gt; >
> > ps. this ENABLEDELAYEDEXPANSION problem, with the exclamation mark, is
> > killing me...
> > ...because I can't get my batch file to read all files without problems..
> > so what I'm doing is, taking the exclamation mark out of the files,
> running
>
on the files manually!
thanks
Clamarc
Em seg., 21 de fev. de 2022 às 22:02, Reino Wijnsma
escreveu:
> On 2022-02-21T08:52:44+0100, CMG DiGiTaL wrote:
> > I did these commands below
>
> This has nothing to with your initial question "Get sample rate in mp3
> files", o
hromaprint
libavutil 57. 7,100 / 57. 7,100
libavcodec 59. 12.100 / 59. 12.100
libavformat 59. 6,100 / 59. 6,100
libavdevice 59. 0.101 / 59. 0.101
libavfilter 8. 14.100 / 8. 14.100
libswscale 6.1100 / 6.1100
libswresample 4.0.100 / 4.0.100
libpostproc 56.0.100 / 56.0.100
*%G: No suc
eally appreciated!
Clamarc
Em dom., 20 de fev. de 2022 às 22:44, Reino Wijnsma
escreveu:
> On 2022-02-21T01:38:30+0100, CMG DiGiTaL wrote:
> > OK, I would also need, along with the sample rate, to get the bitrate of
> > the file, because I realized that in the conversion, because I'
ct
bitrate of each file?
thanks
clamarc
Em dom., 20 de fev. de 2022 às 21:30, Reino Wijnsma
escreveu:
> On 2022-02-21T01:07:12+0100, CMG DiGiTaL wrote:
> > do i need to create the 48000 sample rates folder before running your
> > command, or will it create the folder automatica
atically?
thanks
Clamarc
Em dom., 20 de fev. de 2022 às 19:38, Reino Wijnsma
escreveu:
> Hello Clamarc,
>
> (late reply as I'm a bit behind with cleaning up e-mails)
>
> On 2022-02-07T22:47:29+0100, CMG DiGiTaL wrote:
> > I have a folder "General Audios" and
IMPORTANT^!^
echo.
echo Upon completion of this program, the CONVERTED LUFS
FILES will be on your ^(Desktop^)
echo in the folder named "Converted LUFS Audios" so that you can
move them wherever you want on your PC.
echo After moving the files,
Em sáb., 19 de fev. de 2022 às 14:00, Mark Filipak <
markfilipak.nore...@gmail.com> escreveu:
> On 2022-02-19 08:54, CMG DiGiTaL wrote:
> > hi,
> >
> > I managed to solve the problem of reading several files from this
> code!...
> > ... what is happening now
hi,
I managed to solve the problem of reading several files from this code!...
... what is happening now is that the setlocal EnableDelayedExpansion, is
ignoring files that have an exclamation mark in their name.
I removed the enebladelayedexpansion from the code and changed the format
of the var
hi,
I made a batch file code to change the LUFS of my Audio Files and it is
working and converting the files ok. Turns out it only does this for one
file at a time.
How can I make it convert multiple files at once?... see my code that is
generating one file at a time:
@echo off
chcp 65001
cls
END
t; -v error -show_entries
stream=sample_rate -of default=noprint_wrappers=1:nokey=1 *set* *%%S
if %%S*==48000
("C:\Users\%username%\Desktop\48000 sample rates Audio\%%F")
)
pause
thanks
Clamarc
Em seg., 7 de fev. de 2022 às 16:20, Mark Filipak <
markfilipak.nore...@gmail.com> es
hi Reindl Harald
forgot to send my changed code, follow now, so:
cd\Users\%username%\Desktop\%pasta%
for %%F IN (*) do (ffprobe -i "%%F" -v error -show_entries
stream=sample_rate -of default=noprint_wrappers=1:nokey=1
if %%F==48000 (ffmpeg -i "%%F" -map 0:0 -acodec mp3
-b:a 32
Hi
I made the changes to my .bat file, and the files are not being written for
a detail that I also have doubts about
see how the .bat file was:
Note: I'm using the for command to execute two commands
ffmpeg, okay?
I need one more help from you regarding the If command in
Em dom., 6 de fev. de 2022 às 22:29, Reindl Harald
escreveu:
>
>
> Am 07.02.22 um 02:25 schrieb CMG DiGiTaL:
> > Hi,
> > I'm creating a bat file that will access a folder with several mp3
> files. I
> > would like to be able to select in this folder only the mp
Hi,
I'm creating a bat file that will access a folder with several mp3 files. I
would like to be able to select in this folder only the mp3s with a sample
rate of 48000 khz and copy them to another folder where I will change from
48000 khz to 44100 khz. What command can I use to select only 48000kH
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