Quality also doesn’t matter much, processing speed is more important in my
case
On Thu, Sep 14, 2023 at 3:03 PM Romain Beauxis
wrote:
> Have you tried using libshine instead of lame?
>
> https://github.com/toots/shine
>
> On Thu, Sep 14, 2023, 13:54 Alexander Bieliaev via ffmpeg-
I tried to do like this:
-i "\\.\pipe\FFMpegCore_b9e2a5fa-e574-4c64-a3fa-c5e195fca395" -c:a
libmp3lame -f mp3
"\\.\pipe\FFMpegCore_de5d8dc0-cc91-4d8d-afe5-1240d7b06e9b" -y
doesn't seem to run much faster
am I missing something?
Thanks!
On Wed, Sep 13, 2023 at 7:16 PM H. Vidal, Jr.
wrote:
> O
This is the output:
ffmpeg version 4.4.1-full_build-www.gyan.dev Copyright (c) 2000-2021 the
FFmpeg developers
built with gcc 11.2.0 (Rev1, Built by MSYS2 project)
configuration: --enable-gpl --enable-version3 --enable-static
--disable-w32threads --disable-autodetect --enable-fontconfig
--enab
Hi! I'm using this command to convert wav audio to mp3:
-i "\\.\pipe\FFMpegCore_d1110724-bf07-4ada-9ea4-5b404aea3d52" -b:a 64k -f
mp3 "\\.\pipe\FFMpegCore_c9edc9bd-8061-42a8-a806-ff34bfa7408d" -y
I would like it to go faster. How can I make it faster?
Thanks!
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ote:
> Am 16.02.2023 um 12:31 schrieb Alexander Bieliaev via ffmpeg-user:
> > How can I analyze the volume of input audio (in dB) and write that to the
> > text file?
> >
>
> If you want to know the mean and maximum volume in the whole stream,
> that can be done with the vol
How can I analyze the volume of input audio (in dB) and write that to the
text file?
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I am processing audio chunks programmatically by using the ffmpeg library
for C#. First I divide the input audio of .wav format into chunks of 1
minute each (I can't process the whole audio for specific reasons), then
prepend it's header to each chunk so it can be recognized and processed,
then I g
How can I replace multiple parts of audio with generated beeps?
I tried next command:
-filter_complex "
[0]volume=0:enable='between(t,10,15)+between(t,20,30)'[main];
sine=f=800,
pan=stereo|FL=c0|FR=c0,
volume=0:enable='between(t,0,10)+between(t,15,20)'[beep];
[main][beep]amix=inputs=2:duration=fir
Thanks, it kinda worked for me, but it seems like it's dependent on the
length of the inserted audio. For example if I have a 2 minute audio and
try to replace some parts of it with audio that is 1 minute long, for the
first minute of the audio it works fine but if I try to replace parts above
1 mi
When transcoding audio from WAV and GSM formats to MP3, the result is
shorter in duration than the original. How can you avoid this and keep the
original duration?
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How can I replace a part of the audio from and to a specific time with some
other audio/sound (I want to replace it with beep in this case)?
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I
Hi!
First I wanted to thank you for the wonderful resource you are developing!
I am developing a C#/.NET application for Ubuntu 22.04. The purpose of the
application is to convert gsm610 audio to mp3 (possibly with an
intermediate conversion to wav) without cutting the audio length and
preserving t
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