21
votes
Accepted
Why does ffmpeg's conversion from mono to stereo lower the volume?
1 - It seems that FFmpeg attempts to respect a common pan law (when routing a centered mono track to a stereo track, lower the mono signal by - 3 dB on each channel of the stereo track). The goal is ...
- 2,794
6
votes
Accepted
How to make all sound be at approximately the same level?
Better is subjective, and taste differs, but a chain of effects might include the following:
High pass filter, maybe set at 20Hz. Depends on the material, but if you have a recording done with ...
- 1,354
5
votes
How to change volume of a .wav file in a mp3gain-like way?
ReplayGain tags aren't standard in WAV files, so you have to alter the PCM data with the required gain.
As per my reading of the Replaygain specs, a correctly implemented Replaygain scanner will ...
- 694
4
votes
Accepted
Audio file in shorter in m4a than in PCM
Raw PCM does not store stream properties, so when you ingest a raw PCM stream, ffmpeg assumes a sampling rate of 44100 Hz whereas your source is 48000 Hz. So new duration = old duration x (48000/44100)...
- 694
3
votes
Accepted
How can these wav/mpeg files be the exact same duration—down to the (apparent) millionth of a second?
Digital audio is sampled i.e. made up of discrete chunks, with each chunk representing a small sliver of time. So the duration will be quantized to multiples of 1/sampling frequency i.e. if sampling ...
- 694
3
votes
Accepted
What is the real time from sample number?
OK you are getting confused between frames as in "SMPTE" frames and audio buffers. The frames you are referring to are actually buffers. They are not 'vision' frames. 25fps refers to vision frames.
...
- 8,735
3
votes
Linear loudness normalization in FFMPEG is in fact dynamic
As mentioned in a comment, FFMPEG will revert to dynamic in the conditions specified in the manual:
linear
Normalize by linearly scaling the source audio. measured_I, measured_LRA, measured_TP, and ...
- 255
2
votes
Accepted
Audacity aac export vs ffmpeg
The original memo file is encoded as alac :
$ mediainfo 100.m4a
General
Complete name : Téléchargements/100.m4a
Format : MPEG-4
Format ...
- 2,794
2
votes
Split audio by frequency
Solution with sox:
HZ="2700"
sox inp.wav "$HZ"-.wav lowpass $HZ
sox inp.wav "$HZ"+.wav highpass $HZ
sox -M "$HZ"-.wav "$HZ"+.wav out.wav
- 19
2
votes
How to decide bitrate for converting wav file to opus file with highest quality
You have some recommendations here :
https://wiki.xiph.org/Opus_Recommended_Settings
For 2 channels, Opus at 128 KB/s (VBR) is pretty much transparent.
High enough may be subjective…
- 607
2
votes
How to input noise values in dBFS for ffmpeg silence detect
The trouble is, there's no such thing as 'dB'… well, of course there is but dB on its own is a measure of relative loudness, not absolute.
See https://en.wikipedia.org/wiki/Decibel#...
- 8,141
1
vote
Accepted
Why converting MP3 from 192k to 64k causes samples to protrude
I'm going to take a shot here, I think your problems are all due to summing two identical (or very similar) audio signals to mono, thus increasing the amplitude by 3 - 6 dB (which then protrude/clip ...
- 2,451
1
vote
Split audio by frequency
With Reaper (free evaluation licence, low cost - $60 - usage licence), you can :
Use a 4-band crossover splitter, assign each band to one of 2 outputs (and use the crossover as a 2-band crossover ...
- 607
1
vote
Possible to ffmpeg encode an MP3 media segment for HLS to follow existing MP3?
The Thing Is, you are not perfectly replicating the output of -f hls by doing what you are doing. You are creating discrete mp3's and expecting them to behave the same way as hls mp3's. HLS is ...
- 8,735
1
vote
Accepted
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.".
Try preprocessing live, instead of only postprocessing; that certainly eliminates post.
"...
- 156
1
vote
Accepted
Loudness normalization makes file four times larger
Inspecting the input and output files, I realized that the output was upsampled from 48 kHz to 192 kHz. I disabled that with the option -ar 48000.
- 255
1
vote
Why is exporting a .wav file as .m4a creating a sync issue in audacity?
This is more a workaround than an answer as I did not manage to link the libavformat library to Audacity. So here are some clues using FFmpeg version 4.1 with libavformat.58.20.100.dylib on macOS.
...
- 2,794
1
vote
How to change volume of a .wav file in a mp3gain-like way?
Wav, afaik, has no equivalent soft-gain parameter.
You could simply use Audacity to normalise to zero [so you have a level playing field], then again to -92.8dB... but -92.8 is going to be awfully ...
- 8,141
1
vote
Accepted
How to change channels in audio (from BOTH to LEFT) without re-encoding/conversion?
I don't believe this could be done without batch conversion. What you could do is load all the files into a batch converter and have them run through an effect or process that would produce the ...
1
vote
Understanding how ffmpeg's silenceremove filter correlates to Adobe Audition
I've never used ffmpeg like this, but I had a quick squint at the filter docs, specifically the silenceremover and straight away I noticed the detection system defaults to RMS. Maybe this is where the ...
- 2,451
1
vote
Dynamic amplification of a sound file
There is no magic trick to make something good out of a bad recording.
It will always sound very bad. That is why is it extremely important to get a good recording at first, with plenty of dynamic ...
- 111
1
vote
Using .wav files works fine in DAW while using converted .mp3 files doesn't
Have you tried exporting the .wavs from Bitwig as 24-bit? It may be that for some reason the the 32 bit-depth(float) upsets the FFMPEG conversion.
- 1,336
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