audacity | Audio Editor | Plugin library
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QUESTION
I am currently trying to get a list of all installed applications and would like to build a feature that can launch those.
I'm using these PowerShell commands:
gci HKLM:\SOFTWARE\Microsoft\Windows\CurrentVersion\Uninstall\* | % { Get-ItemProperty $_.PsPath } | Select DisplayName,InstallLocation
gci HKLM:\\SOFTWARE\\Microsoft\\Windows\\CurrentVersion\\Uninstall\\* | % { Get-ItemProperty $_.PsPath } | Select DisplayName,InstallLocation
in conjunction with ConvertTo-Json
in order to get a good stdout I can work with.
Now, this only gives me the InstallPath without any executables.
Is there any easy way to get the main executable of the applications i nthe list?
Expected Result (Name of the key does not matter):
...ANSWER
Answered 2022-Apr-16 at 00:23Like others have pointed out in the comments, there isn't a conventional way of getting the executable paths of certain programs.
To answer your indirect question of building an app launch method, we can make use of a few things. Fortunately for us, PowerShell has a Get-StartApps
cmdlet that produces an output of the current users installed apps:
QUESTION
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:
ANSWER
Answered 2022-Mar-16 at 00:19If you replace sox
with echo
you will see the command you constructed is not really permutating the way you want:
QUESTION
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.
AudacityI 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
...ANSWER
Answered 2022-Feb-24 at 04:59The numbers 56
..etc refer to the version number of the individual libraries that are a part of the FFmpeg project. So, avcodec-59
is major version 59 of libavcodec.
Yes, you can have parallel installations without issue, provided you can set custom search paths for the client application.
I suspect these clients are using libavcodec to export to major codecs. Those are generally mature, so older versions should be fine.
That's correct. Across major versions, the API and ABI will likely have changed.
Depends on what API calls they're using. A new decode/encode API was introduced some years ago but the old one was kept for compatibility, so clients could transition over gradually. In v59, those old calls were removed. So, if they did transition, then yes.
QUESTION
I want to remove completely silence parts from wav files with ffmpeg.
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
...ANSWER
Answered 2022-Feb-09 at 03:49It looks like the equivalent command to AUdacity's truncate silence behaviour is the following (in bold what is added):
ffmpeg -i input.wav -af silenceremove=start_periods=1:stop_periods=-1:stop_duration=0.2:start_threshold=-45dB:stop_threshold=-45dB output.wav
I am not sure why adding those 2 parameters leads to the expected behaviour but it works although for some files silenceremove can remove more parts than Audacity / silencedetect detect.
QUESTION
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.
...ANSWER
Answered 2022-Jan-17 at 19:06Ok this is awkward, i struggled for days to find the answer so i decided to ask a question. I just found a way to solve it.
- select the point you want to add a break.
- drag across multiple tracks to select them. (did not you you could do that)
- Shift-K to select everything after it.
- drag to the right.
I am sorry for this but i think it may help someone.
QUESTION
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
...ANSWER
Answered 2021-Dec-14 at 12:42The individual interfaces corresponding to a class can be detached by slightly modifying the above code
QUESTION
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.
...ANSWER
Answered 2021-Nov-25 at 20:55If TIFF parlance, you have a file in contiguous planar configuration, and want separate planar configuration.
The tiffcp
utility that comes with LibTIFF can do this for you. Use the -p separate
option:
QUESTION
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.
...ANSWER
Answered 2021-Sep-18 at 12:04This is not a question of "trust". Both functions do what they are supposed to do. The TF version apparently does not resample the audio. Librosa, by default, resamples to 22,050 Hz (for whatever reason). Please read the docs. You can avoid this by calling
QUESTION
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:
...ANSWER
Answered 2021-Jun-04 at 09:07The problem
Your .wav file is Signed 16 bit Little Endian, Rate 44100 Hz, Mono
- which means that each sample in the file is 2 bytes long, and describes a signed amplitude. So you can copy-and-paste chunks of samples without any problems, as long as their lengths are divisible by 2 (your block size). Your silences are likely of odd length, so that the 1st sample after a silence is interpreted as
QUESTION
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)
...ANSWER
Answered 2021-May-07 at 19:52Calculating dB from audio samples isn't that simple. It sounds like what you want to do is find an appropriate threshold and then clip the audio with something like:
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