audacity | Audio Editor | Plugin library

I am trying to combine permutations of some .wav files.
There's 6 variations of 4 instruments. Each generated track should have one of each instrument. If my math is right, there should be 24 unique permutations.
The files are named like:

I have the latest release of FFmpeg, version 5.0-full_build-www.gyan.dev, which is a compiled executable for Windows. I downloaded "ffmpeg-release-full-shared.7z," which includes dlls in \bin.

I have the latest release of Audacity, 3.1.3, which I installed using the 64-bit installer.

In Audacity's Preferences > Libraries, there is an option to use FFmpeg's libraries. Under Help > Diagnostics > Show log, Audacity seems to be looking for one of three versions of avformat-NN.dll: avformat-58.dll, avformat-57.dll, or avformat-55.dll

I want to remove completely silence parts from wav files with ffmpeg.

Input wav can be like :

I am using the following ffmpeg command to remove silence part ffmpeg -i input.wav -af silenceremove=stop_periods=-1:stop_duration=0.2:stop_threshold=-45dB output.wav because I understand from the doc that it will remove all silence parts longer than 0.2 s (silence being below -45dB).

But I get that where silence part has only been reduced to around 0.1 wheras I want it to be 0 (no remaining silence).

In Audacity I will use "truncate audio" filter and choose the above parameters to detect silence and in the action part I will choose to truncate to 0: .

This will yield to what I want (ie an audio with no silence part remaining):

Searching on the internet only lead me to what I already do.

So how can I reproduce the output I get from Audacity "Truncate Silence" filter with ffmpeg and remove all silence parts from audio ?

Edit: The output from silencedetect filter is correct: ffmpeg -i input.wav -af silencedetect=0.2:n=-45dB -f null - detects exactly what audacity detects.

Thanks in advance for your help

I made a Audacity Project with multiple tracks and multiple clips on these tracks. Now I want to add a pause at a point but I can not figure out how. Moving every clips on every tracks after this point manually would be really painful, is there an alternative? I am using the latest version of Audacity on Windows 10.

I hope this is not a wrong place for Audacity.

I am currently working on a project involving composite USB Device, I am using STM32Fxx for the process. One of the classes is USB Audio Class, which works perfectly(tested with audacity).

Due to my requirements I need to control the audio streaming with pyusb. I have already detached the drivers attached to the device using the following

I have been trying to figure out a way to create non-interleaved .tiff files, as described here: https://questionsomething.wordpress.com/2012/07/26/databending-using-audacity-effects/ (under the heading of "The photographic base").

It seems like it's a trivial thing using photoshop, but I'm on linux and would hate to get myself a copy just for this one option. If anyone knows of a way, be it via imagemagick, hacking the gimp or some obscure program, I'd be glad for any suggestions.

I have one wav file which I resampled to 16.000 kHz with Audacity. Now I am trying to load the file with python with 2 different ways.

I am currently trying to make a .wav file that will play sos in morse.

The way I went about this is: I have a byte array that contains one wave of a beep. I then repeated that until I had the desired length. After that I inserted those bytes into a new array and put bytes containing 00 (in hexadecimal) to separate the beeps.

If I add 1 beep to a WAVE file, it creates the file correctly (i.e. I get a beep of the desired length). Here is a picture of the waves zoomed in (I opened the file in Audacity): And here is a picture of the entire wave part:

The problem now is that when I add a second beep, the second one becomes completely distorted: So this is what the entire file looks like now:

If I add another beep, it will be the correct beep again, If I add yet another beep it's going to be distorted again, etc. So basically, every other wave is distorted.

Does anyone know why this happens?

Here is a link to a .txt file I generated containing the the audio data of the wave file I created: byteTest19.txt

And here is a lint to a .txt file that I generated using file format.info that is a hexadecimal representation of the bytes in the .wav file I generated containing 5 beeps (with two of them, the even beeps being distorted): test3.txt

You can tell when a new beep starts because it is preceded by a lot of 00's.

As far as I can see, the bytes of the second beep does not differ from the first one, which is why I am asking this question.

If anyone knows why this happens, please help me. If you need more information, don't hesitate to ask. I hope I explained well what I'm doing, if not, that's my bad.

EDIT Here is my code:

I am dealing with an audio file that shows sudden spikes:

When I try to normalize the audio file, the audio program sees this spike, notices that it is at 0 dB and won't normalize the audio any more.

To solve this issue, I have applied a Limiter that limits to -3 dB. I have used both Steinberg Wavelab and Audacity Hard Limiter. Instead, they both diminish the entire audio volume.

Both do not eliminate this spike.

Edit: I have found out that "Hard Clip" in Audacity does what I need, but now I still want to finish my own approach.

So I was thinking that they perhaps do not work correctly.

Then I tried to write my own limiter in VB6 in order to have full control over what's happening.

To do that, I'm loading the audio data of a wav file like this:

(I have stripped the process down very much)