Questions tagged [ffmpeg]

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Record live audio aka Stereo Mix in WIndows 10 or 11 with ffmpeg - Windows WASAPI

Long story short it seems that StereoMix heavily used in past was a feature of the sound card and sound drivers, so... it's not possible to use it if it isn't present on the machine. C:\Users\user\...
user3450548's user avatar
1 vote
1 answer
180 views

Why converting MP3 from 192k to 64k causes samples to protrude

It doesn't matter which software is used to convert, FFMPEG or CoolEdit 2000, it makes always the same effect. The original file is WEBM. When I convert it to WAV PCM or MP3 192k using one of ...
Paul's user avatar
  • 113
1 vote
1 answer
189 views

Audacity aac export vs ffmpeg

Apologies if this is off topic here. I have dozens of small audio files recorded using iOS VoiceMemo. These files fail to load properly in a web-based experiment that I am trying to run. When I open ...
Alan Munn's user avatar
  • 113
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0 answers
258 views

How can I convert from PCM to EVS audio file and decode back to PCM?

everybody. I have a problem with converting to 3GPP TS 26.442. If I want to convert from .48k to .192 file, how am I supposed to get original audio file? Do I have to convert from any audio file (eg. ...
MEVSA12's user avatar
-1 votes
2 answers
513 views

Split audio by frequency

I would like recode wav files from stereo to 4 channels - front channels with low frequencies and back channels with high frequencies only; to store wav files on DLNA server and play on receiver in bi-...
user1855805's user avatar
1 vote
1 answer
626 views

Possible to ffmpeg encode an MP3 media segment for HLS to follow existing MP3?

My use case is essentially DIY internet radio. I've abstracted the real-world constraints into a toy problem for simplicity, but these constraints are real for me. I'm shipping a 24/7 live HLS stream ...
Charney Kaye's user avatar
0 votes
0 answers
603 views

Can the ffmpeg `loudnorm` extension be used to guarantee a maximum volume in an audio file?

I'm looking at a way to ensure an audio file doesn't ever get too loud, but I do not want to change the samples if it is under a given threshold. For example, I would like the file to have a maximum ...
Alexis Wilke's user avatar
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0 answers
40 views

How to create a sense of motion in stereo audio

I have a :15 mono audio file of a car driving by. What I would like to do is use that as the input to create a stereo audio file where the sound 'moves' from left speaker to right for better realism. ...
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0 votes
1 answer
481 views

How can I improve on my post processing using ffmpeg?

My goal is to record a video and then upload it without any further manual post production. Except for the sound which I can do with the push of a button using ffmpeg. So far I managed to remove the ...
greeven's user avatar
  • 11
1 vote
1 answer
403 views

How to decide bitrate for converting wav file to opus file with highest quality

ffmpeg -i file.wav -ab 320k newfile.opus I don't know how to decide the value to set for the -ab option. But my overall goal is to convert wav file to opus file with high quality, and delete the wav ...
qazwsx's user avatar
  • 111
2 votes
1 answer
921 views

How to input noise values in dBFS for ffmpeg silence detect

I used the silencedetect of ffmpeg with noise value of -30dB to get the silent parts of a small video file & used those timestamps to trim(remove) & re-encode the silent parts of the video. I ...
icyGuy's user avatar
  • 21
1 vote
1 answer
297 views

How to make all sound be at approximately the same level?

I tried to use normalize (in Audacity and in ffmpeg), but it seems there is still room for improvement. Is it possible to do something better than normalize?
R S's user avatar
  • 127
2 votes
1 answer
2k views

Linear loudness normalization in FFMPEG is in fact dynamic

I followed this post to normalize the loudness of a clip. I first measure the loudness: $ ffmpeg -i "Take109 000-012.wav" -filter:a loudnorm=I=-24:TP=-9:LRA=5:print_format=summary -f null - ... [...
miguelmorin's user avatar
2 votes
1 answer
982 views

Reasonable compressor settings for radio and podcasts

I am moving from software engineering to audio production. I read a good deal about audio editing (mostly from Transom.org) and I want to produce audio for both radio at -24 LUFS and podcasts at -16 ...
miguelmorin's user avatar
0 votes
1 answer
216 views

Loudness normalization makes file four times larger

I have several WAV files that I pass through FFMPEG's loudness normalization: ffmpeg -i input.wav -filter:a loudnorm=I=-24:TP=-9:LRA=7 output.wav The output files are on average four times larger (4....
miguelmorin's user avatar
1 vote
0 answers
250 views

How to use a jack route as a single audio-filter with ffmpeg

I have configured Ardour and there, An audio track takes input from a Jack-Sink device Applies some effects by using some Aux-Send Outputs the final audio to Master which is a Jack-Client Now, I ...
Chitholian's user avatar
1 vote
0 answers
127 views

ffmpeg silenceremove on 2 track audio

Is there any way to use silenceremove on file with 2 audio tracks, to trim silence only when the silence is on both audio tracks? File info: .mp4 with 2 stereo audio tracks(aac) I've already tried ...
Quiet Taxi's user avatar
3 votes
1 answer
1k views

What is the real time from sample number?

I apologize if this is the incorrect forum for my question, so please point me to the right place if it doesn't belong here. I thought I understood how SMPTE timecode works, but I cannot seem to ...
HonestMath's user avatar
2 votes
1 answer
849 views

Why is exporting a .wav file as .m4a creating a sync issue in audacity?

Edit: it's 0.0330s delay, not half a second sorry. ALso the problem occurs when converting the wav to .m4a in ffmpeg using the basic ffmpeg -i input.wav output.m4a, and I used the newest release of ...
kite's user avatar
  • 153
1 vote
1 answer
1k views

Audio file in shorter in m4a than in PCM

I recorded a voice memo that shows a length of 10:22 in iTunes. I convert it to PCM with ffmpeg -i input.m4a -c:a pcm_s16le -f s16le temp.pcm and then back to m4a with ffmpeg -f s16le -i temp.pcm -...
miguelmorin's user avatar
0 votes
1 answer
70 views

How can these wav/mpeg files be the exact same duration—down to the (apparent) millionth of a second?

I'm working on a project that involves many different .wav files. In checking out some attributes of these files, I noticed something strange: The duration of all the files looks to be chunked in some ...
AmagicalFishy's user avatar
0 votes
2 answers
3k views

How to change volume of a .wav file in a mp3gain-like way?

Well, i have a bunch of WAVE files for a Fallout 3 mod, but in the mod instructions the author only explains how to accomplish it using mp3gain (and of course for .mp3 files). The only info it have is ...
Vico's user avatar
  • 101
0 votes
1 answer
229 views

How to change channels in audio (from BOTH to LEFT) without re-encoding/conversion?

lets say I have very long audio file, and I want to remove only right channel from BOTH (stereo) channels. Can that be done without re-encoding, just simply switching off one channel? (maybe FFMPEG??)...
T.Todua's user avatar
  • 141
12 votes
1 answer
8k views

Why does ffmpeg's conversion from mono to stereo lower the volume?

We start with this mono file: test.wav In audacity it looks like this: Then we convert it to stereo with this command: ffmpeg -i test.wav -ac 2 out.wav Now it looks like this: Notice that the ...
Igor Liferenko's user avatar
5 votes
2 answers
2k views

Repair audio file without header

We recorded the audio of a rather long meeting of a couple of hours on a phone. Someone forgot to turn off sound recording at the end of the meeting, and the phone must have turned off due to low ...
just.kidding's user avatar
2 votes
1 answer
3k views

Understanding how ffmpeg's silenceremove filter correlates to Adobe Audition

I have a bunch of audio files (more than 4000) that I want to trim the leading and trailing silence from, that's where ffmpeg comes in. For instance in the image bellow, the command I'm using is $ ...
rraallvv's user avatar
  • 345
2 votes
2 answers
564 views

Dynamic amplification of a sound file

I have a 60-minute WAV file of a conversation. Here is a 600kB clip from it: https://www.dropbox.com/s/m00d525v34bwbpa/needs-levelling.wav?dl=0 One voice is very quiet and I would like to amplify it....
ConvexMartian's user avatar
0 votes
1 answer
300 views

Using .wav files works fine in DAW while using converted .mp3 files doesn't

I am working on Arch Linux and am trying to create an intro movie for my Youtube channel. The idea is to play "intro.mp3" which is played only once and then "loop.mp3" which will be playing on low ...
71GA's user avatar
  • 121
1 vote
0 answers
1k views

Intelligent downmix of stereo to mono

Is there a technique for intelligently downmixing stereo to mono with FFmpeg or similar? As we know, a signal phased between the left and right channels can cancel each other out when mixed together. ...
Brad's user avatar
  • 2,588