mysteryx93/hanumaninstituteapps

432Hz Player, 432Hz Batch Converter, Powerliminals Player, Yang YouTube Downloader

hanumaninstituteapps's Issues

432Hz Batch Converter - When advanced settings window is open, it doesn't show up in alt-tab

Tested with the AppImage on Linux Mint 20.3 that has been fully updated as of 2023-05-07

Simply put, in the 432Hz Batch Converter, if you open the "Advanced" settings window and then change focus to a different program window (e.g. Firefox) and then try to alt-tab back to the 432Hz Batch Converter, the 432Hz Batch Converter will not be shown in the alt-tab list as long as that "Advanced" settings window is open.

But as long as the "Advanced" settings window is closed, then the 432Hz Batch Converter will once again show up within the alt-tab menu:

[Enhancement]　432Hz Batch Converter - Option to instead alter playback sample rate and directly copy source audio stream

As mentioned in issue #36, there are times when you're not able to use a transcoded version of your source audio and therefore are forced to instead directly alter the playback sample rate.

It would be great if 432Hz Batch Converter had the functionality to directly write that sample rate value (rounding to the nearest integer of course) to what is nothing more than a copy of the source audio stream, making it a lossless operation without transcoding (and obviously this would by definition be locked to only be able to do "skip tempo adjustment for quality" since anything else isn't possible without transcoding).

　
Now this obviously won't work for the mentioned proprietary ADPCM formats, but it can work for more standard formats, particularly WAV and OGG vorbis (the latter of which is used a lot in games and visual novels).

To clarify, there's an old WinXP-era program called "Header Investigator" (it even works in wine) that can do this for WAV files, but it's limited to specific values unless you hex-edit the program EXE (which I did previously to add support for 32bit as well as 176400Hz):

• http://www.railjonrogut.com/HeaderInvestigator.htm
• https://hydrogenaud.io/index.php/topic,119735.msg994544.html#msg994544

　
However, I don't believe there's any software that does this for OGG vorbis files let alone any other possible formats like at least M4A/AAC and FLAC (MP3 and Opus only support a specific set of sample rates).

It's important to note that, at least for OGG vorbis, changing the hex sample rate value results in a CRC error that not all software will handle (foobar2000, MPC-HC, and VLC can't read said OGG vorbis files, but Audacity, Audacious, and Celluloid/mpv can read them without issue).
I did discover however that if you first package your source unmodified OGG vorbis into an mkv, hex-edit the sample rate value, and then re-extract the vorbis stream back into an OGG container, it'll avoid the CRC error issue (though some vorbis streams end up being only like 99.999% lossless when doing this for some reason, having a couple single-sample -50dB deviations oddly enough).

Converter432Hz no longer opens any files-- just updated...

Hello--- I LOVE your software-- I use it daily, multiple times daily. I think the program has an issue with it's storage section or GUI something like that-- can't get any log files-- the only thing that is output is in Appdata/Roaming --- Ads.json and 432HzConverterConfig.json --- but I did see Avalonia-- and MvvmDialogs mentioned over and over in the error-- also--- this never happened on v3.2 --- happens on any computer I put it on that is running windows 10. I'm on Windows 10 by the way--- I did try it on opening files from my main c drive-- and it worked so just a heads up if that helps anything out. never commented much on anyones software before-- but yours was worth it-- wonderful software--- keep up the good work, thank you.

432Hz Batch Converter - FLAC @ 32bit exports at 16bit rather than 32bit integer

Simply put, setting the output format to FLAC with 32bit results in the output being 16bit instead of the expected 32bit integer (which recent versions of the FLAC encoder finally added support for).

At the very least I would think that it should use 24bit rather than 16bit if it cannot support 32bit (better yet would be to dynamically remove 32bit from the drop-down list when selecting FLAC, resetting any preexisting 32bit selection to 24bit instead)

Crashes while startup

Works great on my Computer, but crashes while startup on my laptop.

• Doubleclick .exe
• after 3-5s a Window pops up without content
• after 3-5s windows closes without any error

Any ideas?

The file C:\Natural Grounding\Job1_Output.avi was not found

I'm trying to convert a video using Yin Media encoder but always have this error:
The file C:\Natural Grounding\Job1_Output.avi was not found.

PS: I even tried convert to avi but got same error

something error with YinMediaEncoder

Yin Media Encoder: Cannot convert 2160p video?

Hi,

I've really appreciated your converter over the past few months, but yesterday, I had some unknown problem. As soon as FFMPEG launches, it gets this far before hanging there. I left it on overnight (~12 hours) with no result.

Here's the output:
ffmpeg-output.txt (I had to upload it in a text file as Markdown seems to break line breaks in code.)

And that's it. I do have a process called "avs2yuv.exe" running in the background, but it stays locked at 1.6 MB memory usage, 0% CPU and 0 MB/s disk usage.

The output file in C:\Natural Grounding\Temp stays locked at 0 bytes the entire time. I have attached the script and settings from that folder here: Job11.zip

I have also attached a MediaInfo output of the original file here:
MediaInfo.txt

Thanks in advance,

HewwoCraziness

432Hz Batch Converter - exporting as WAV causes any preexisting metadata to be removed

Simply put, if the output format is set to WAV, then any metadata that existed in the source audio will not be present in the resulting WAV file.

This is in contrast to a program like foobar2000 which retains the metadata when converting and exporting to a WAV file.

By comparison, the metadate is still present if you select FLAC, but WAV is (currently) the only way to get a lossless 32bit output so that you can subsequently do additional audio processing (e.g. ReplayGain, WaveGain, r128gain, etc) without quality loss.

[Enhancement]　432Hz Batch Converter - export option to additionally apply r128gain／ReplayGain／WaveGain

Currently any r128gain／ReplayGain／WaveGain needs to be applied separately before or after conversion. Ideally the more modern r128gain algorithm should be used, but even the older original "Replay Gain" algorithm is better than nothing.

Since the dynamic range of some more classical songs can have an average perceived volume that is lower than the default 89dB even when the waveform is peak-normalized, it'd be best if it were a configurable setting what dB should be targeted.

Regardless, I envision three sort of modes:

• apply directly to the waveform (WaveGain)
• apply only metadata (r128gain／ReplayGain)
• peak-normalize the waveform and only apply metadata

The last one is designed to maximize your bit depth resolution, particularly if you're limited to 16bit for whatever reason.

For the most accurate volume adjustments, any r128gain／ReplayGain／WaveGain should be applied at the very end of the processing chain, that being the step immediately before the audio is actually encoded into your selected output format (e.g. MP3, AAC, FLAC, WAV, etc).

Speaking of WAV, the metadata function might not work specifically when exporting WAV files unless issue #29 were first resolved.

EDIT: Just re-discovered that this would also alleviate the quirk caused by resampling where 432Hz Batch Converter can end up clipping the resulting waveform a bit, though that itself could be possibly considered a stand-alone bug.

Auto-Detect Exact Pitch

See if we can auto-detect exact source pitch.

It has been implemented here in Python.
https://github.com/CardLin/Exact432HzConverter

conversion repeatability

hello, if an mp3 file has been converted to 432, why is a conversion of the converted one not identical to itself?

432Hz Batch Converter - Doesn't seem to do proper floating point decoding or something

Tested with the AppImage on Linux Mint 20.3 that has been fully updated as of 2023-05-07

Simply put, the 432Hz Batch Converter doesn't seem to be properly decoding floating point or something - it might be truncating everything to 24bit integer and/or incorrectly treating 32bit floating point as 32bit integer instead.

The two main tests I did were to input all 4 of the attached files and have the output set to 32bit WAV and a sample rate of "source"; the advanced settings were configured as:

If you then open the original normalized.wav as well as the "converted" normalized.wav file in Audacity in a single project, invert one of the waveforms, and then mix the two together and then do an Amplify, you'd expect Audacity to report "infinity" but... it doesn't, and instead will let you amplify without clipping by around 90dB or so.

On the other hand, if you open the "converted" wav.wav, aac.wav, and opus.wav files in an Audacity project and do an Amplify, you'll see that it'll say 0dB. This very much implies that floating point data is not being decoded because, if you do the exact same Amplify process with the source wav.wav, aac.m4a, and opus.opus, you will see that Audacity reports that you can "Amplify" by something like negative 48dB.

　

EDIT: This is hilarious - github lets you attach MOV, MP4, and WebM files, but not WAV, M4A, nor Opus? Seriously...

So here's the next-best thing - a ZIP archive containing the aforementioned audio test files hosted on mpv's go-to temporary file host:

• http://0x0.st/HZ6v.zip

Feature Request: 528hz encoding

Hi,

I like the idea of having 528hz instead of 432hz. Could you please add it? Thank you!

Also, I wrote this small web app: (I thought you might be interested) I hope its not too "off-topic"

https://zen-wood-chime-generator.sourceforge.io/index.html

Additional features

Some features that could be added in the future

• Drag-drop support
• Command-line support
• Support running on server without sound card?

How to use this repo to download live stream m3u8 URL and save to local hard drive?

Hello:
I happen to find your repo, and it seems to be interesting. However, as it is rather complicated,
I can’t figure out yet how to download multiple M3U8 videos and save to local hard drive.
For example, I have the following 2 M3U8 videos in the URL:
http://demo.unified-streaming.com/video/tears-of-steel/tears-of-steel.ism/.m3u8
http://devimages.apple.com/iphone/samples/bipbop/bipbopall.m3u8
I want to launch 2 clients to download them at the same time, when finished, I want to save the files as local files: C:\1.mp4 and C:\2.mp4 and stop or dispose the client.
Please show me how I can do this with C#.
By the way, I am using Windows 10 Version 21H1, and Visual Studio 2019 Version 16.10.4.
Thanks,

432Hz Batch Converter - "Shift pitch" text fields max out at 4 digits (technically 5 if you include a value of 10000)

As you know, interpolation and high frame rates can be quite useful, and it's best to use exact multiples e.g. 2x, 3x, etc.

However, many TVs, such as my own, does not support any refresh rates below 49.091Hz nor any refresh rates above 64Hz.

So for 24fps content, I will sometimes instead slightly speed them up to 24.55fps so that, with 2x interpolation, they'll play at 49.1fps.

This is easy enough with true 24.000fps content since the ratio necessary only needs 3 digits for exact precision—with "round pitch" unchecked and "skip tempo adjustment for quality" checked, you simply just set it to shift pitch from 480Hz to 491Hz.

The problem are those pesky 24000/1001 videos (more commonly referred to as 23.976) since the exact ratio for that would be shifting pitch from 480000Hz to 491491Hz which requires 6 digits of precision, and 432Hz Batch Converter only allows for 4 digits of precision (technically 5 if you include a value of 10000Hz).

Making the json settings file read-only causes apps to crash

Hi, I love the apps you built! However, I noticed that if I make the json preferences file read-only, the apps crash when I try to run them.

So I wanted to make the configuration file read-only so that I could start the app normally, select the settings I need, and then set custom default settings by making the configuration file read-only. That way, I don't have to worry about forgetting to revert back, since the other settings are temporary and always revert back to the custom defaults.

I am using Windows 10.

LOL I forgot to say which apps the crashes occur in! But after further testing, the crashes seem to be happening in all four apps.

Message=Field not found: 'Avalonia.Controls.Presenters.ItemsPresenter.AreVerticalSnapPointsRegularProperty'.

System.MissingFieldException
HResult=0x80131511
Message=Field not found: 'Avalonia.Controls.Presenters.ItemsPresenter.AreVerticalSnapPointsRegularProperty'.
Source=FluentAvalonia
StackTrace:
在 CompiledAvaloniaXaml.!AvaloniaResources.XamlClosure_626.XamlClosure_627.Build(IServiceProvider )
在 Avalonia.Markup.Xaml.XamlIl.Runtime.XamlIlRuntimeHelpers.<>c__DisplayClass1_0`1.b__0(IServiceProvider sp)
在 Avalonia.Markup.Xaml.Templates.TemplateContent.Load(Object templateContent)
在 Avalonia.Markup.Xaml.Templates.ControlTemplate.Build(TemplatedControl control)
在 Avalonia.Controls.Primitives.TemplatedControl.ApplyTemplate()
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Controls.StackPanel.MeasureOverride(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Controls.Grid.MeasureCell(Int32 cell, Boolean forceInfinityV)
在 Avalonia.Controls.Grid.MeasureCellsGroup(Int32 cellsHead, Size referenceSize, Boolean ignoreDesiredSizeU, Boolean forceInfinityV, Boolean& hasDesiredSizeUChanged)
在 Avalonia.Controls.Grid.MeasureCellsGroup(Int32 cellsHead, Size referenceSize, Boolean ignoreDesiredSizeU, Boolean forceInfinityV)
在 Avalonia.Controls.Grid.MeasureOverride(Size constraint)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Controls.Grid.MeasureOverride(Size constraint)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Layout.LayoutHelper.MeasureChild(Layoutable control, Size availableSize, Thickness padding, Thickness borderThickness)
在 Avalonia.Controls.Presenters.ContentPresenter.MeasureOverride(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Layout.LayoutHelper.MeasureChild(Layoutable control, Size availableSize, Thickness padding)
在 Avalonia.Controls.Decorator.MeasureOverride(Size availableSize)
在 Avalonia.Controls.Primitives.VisualLayerManager.MeasureOverride(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureOverride(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Layout.Layoutable.MeasureOverride(Size availableSize)
在 Avalonia.Controls.Window.MeasureOverride(Size availableSize)
在 Avalonia.Controls.WindowBase.MeasureCore(Size availableSize)
在 Avalonia.Layout.Layoutable.Measure(Size availableSize)
在 Avalonia.Layout.LayoutManager.Measure(Layoutable control)
在 Avalonia.Layout.LayoutManager.ExecuteInitialLayoutPass()
在 Avalonia.Controls.Window.ShowCore(Window owner)
在 Avalonia.Controls.Window.Show()
在 HanumanInstitute.Avalonia.CommonWindow.<>n__0()
在 HanumanInstitute.Avalonia.CommonWindow.d__1.MoveNext() 在 D:\HanumanInstituteApps\Src\Avalonia\Controls\CommonWindow.cs 中: 第 27 行

Feature Request: Internet Radio Stream support for player.

It would be a great help if you could also play internet radio streams.

Thank you¡

432Hz Batch Converter - does not prevent waveforms from clipping

If you have a waveform that is close to peak 0+-dB then there's a decent chance that, after running it through 432Hz Batch Converter, the resulting output will actually be clipped (especially if resampling to substantially higher sample rates).

I have a particular waveform that I had to de-amplify to have a peak of around 1.2dB before running it through 432Hz Batch Converter in order for the result to not be clipped, especially compared to a copy of the same waveform that was normalized.

Here's the normalized sample of it (encoded as 32bit WavPack):

• http://0x0.st/HlaN.wv

I was able to use the Replay Gain scanner in foobar2000 in order to measure the average loudness whereby I could put hard numbers to seeing that there was about a full 1dB difference of perceived volume between the two waveforms even though, if I normalized both of the resulting waveforms, there'd only be like 0.1dB or 0.2dB of a difference in peak, implying that the originally-normalized version has about 1dB of its waveform peaks clipped.

Alternatively, if I output the normalized waveform to M4A/AAC and then do an "amplify" process in Audacity, then I can see that there's a bit under 1dB of the waveform that's being clipped but is still preserved via the format's floating-point nature.

(normally when outputting to 32bit WAV you should be able to similarly de-amplify the resulting waveform and recover the clipped data, but with that format the clipped data is missing which is why I previously created issue #17)

Ideally it'd probably be best to have an on/off toggle setting labeled something like prevent clipping which simply reduces the gain of any waveform that would ordinarily clip until it no longer clips.

linux convert with ffmpeg

hello, do you think it could work well?

---

#!/bin/bash

Batch convert all .mp3 files in the current directory to 432Hz with ffmpeg

Options

suffix="false" # Append the -432Hz suffix or not

oldIFS=$IFS
IFS=$'\n'
set -f # Deal with blanks and special characters in file names of the file command and in for loop

found=($(find . -name "*.mp3")) # Find the files in the current directory

IFS=$oldIFS
set +f
filename=""

for file in "${found[@]}"; do # Iterate the found files

Math:

We want to convert from 441Hz to 432Hz, the difference is 9Hz which are 2,040816327 % of 441Hz.

Parameters:

asetrate contains the desired pitch (frequency) but changes the playback speed

atempo ajusts the resulting playback speed, which should remain the same

aresample keeps the pitch change by correct rate of (re)sampling

mv "$file" "$file.tmp"

if $suffix == "true"; then
filename="${file%.mp3}-432Hz.mp3"
else
filename="$file"
fi

ffmpeg -loglevel 8 -i "$file.tmp" -af asetrate=43200,atempo=1.00,aresample=43200 "$filename"
rm "$file.tmp"
echo "Pitched $filename"
done

[Enhancement]　432Hz Batch Converter - allow exporting to WavPack (at least in lossless)

Simply put, have WavPack as an exportable format, at least the lossless variant of the format.

The main thing is that WAV files are the only format that currently support 32bit floating point (though, as previously stated with issue #17, it might be slightly bugged) but WAV files only support files up to 4GB (which is a limit I've run into when working with longer, high sample rate files)—WavPack files however are very much not limited to 4GB and also support 32bit floating point.

It may be worth noting that, unlike formats like FLAC which are designed with a simple single value for how compressed a file should be with the trade-off of higher CPU usage (only really a concern for old sub-GHz CPUs), WavPack has not only a similar sort of "compression vs speed" value but also an "additional processing* value which can be used to compress the file even farther but only impacts the encode speed and not decode speed.

Now WavPack also supports hybrid-lossy, but this was never something that really "caught on" for a multitude of reasons, so I question if it'd even be worth spending the time and effort to make the software and its interface be able to handle encoding into WavPack's hybrid-lossy mode rather than just only supporting lossless WavPack.

Question - does purchasing a license provide any perks other than ad free?

I'm just wondering if purchasing a license provides anything more than the "warm fuzzies" of knowing that you're helping the projects development, i.e. if it also gives you access to priority support or the like.

"Skip tempo" by itself gives different waveform length than "Skip tempo" + "Round pitch"

This discrepancy I previously reported a couple of months ago back in the beta version(s) seems to still exist.

Simply put, having Skip tempo adjustment for quality checked but Round pitch for quality unchecked results in a waveform that has a different length compared to if you have Skip tempo adjustment for quality checked and also Round pitch for quality checked.

...well, at least with the AppImage on Linux Mint 20.3 Xfce.

Nevertheless, the point is, why is there a difference at all between these two configurations? I thought the benefit of Round pitch for quality was for when Skip tempo adjustment for quality was unchecked, but maybe my impression was wrong...?

Yin Media Encoder not recognized

After add source video to Yin Media Encoder show problem with identification source video.

Support for HEVC / x265 source files

The YinMediaEncoder works as intended for any file that is not encoded with x265. Those files simply produce a green output file - no picture (sound gets passed through).

Could support for hevc files be added so we can frame interpolate them as well?

432Hz Batch Converter - OGG/Vorbis format output is limited to 48kHz sample rate despite the format supporting higher

Simply put, OGG vorbis is able to be encoded into much higher sample rates than 48kHz (I know in the past with other audio conversion software I've successfully output 192kHz OGG vorbis, and no I'm not mixing it up with 192kbps) yet 432Hz Batch Converter limits its output for that format to 48kHz.

This isn't something related to it being a lossy format either because 432Hz Batch Converter allows up to 96kHz output for M4A/AAC audio.

Error reading the files (Linux)

Field report:
I have a directory in which there are 145 directories with mp3 files. (approx. 80 GB, 6 ... 8.000 files)
But after the first run of "432Hz" there are only 89 directories in the target directory! I have to run "432Hz" again and let it convert the same directory again. Now it also finds the other 56 directories.
There seems to be a problem reading in the files.
It is also unusual that the reading is not done by the file name but by the timestamp.

Suggested change for " 432Hz":

Files list box (below):
In the application, it would be helpful to be able to change the sort order of the processed files. Now the first edited file is always shown at the top and others run down the list until the screen is over. But it would make sense if the edited files are always shown at the top of the list and then slide down. Limiting the list to 50 ... 100 entries would also make sense.

Thanks for your work.

Translated with www.DeepL.com/Translator (free version)

Fehler beim Einlesen der Dateien (Linux):

Erfahrungsbericht:
Ich habe einen Verzeichnis in dem 145 Verzeichnisse mit mp3-Dateien liegen. (ca. 80 GB, 6 … 8.000 Dateien)
Nach dem ersten Durchlauf von „432Hz“ befinden sich im Zielverzeichnis aber nur 89 Verzeichnisse! Ich muß „432Hz“ noch mal starten und das selbe Verzeichnis erneut umrechnen lassen. Jetzt findet er auch die anderen 56 Verzeichnisse.
Es scheint ein Problem beim Einlesen der Dateien zu geben.
Ungewöhnlich ist auch, dass das Einlesen nicht nach dem Dateinamen sondern nach dem Zeitstempel erfolgt.

Änderungsvorschlag für „ 432Hz“:

Listenfeld Dateien (unten):
Bei der Anwendung wäre es hilfreich, wenn man die Sortierreihenfolge der bearbeiteten Dateien ändern kann. Jetzt wird oben immer die erste bearbeitete Datei angezeigt und weitere laufen nach unten in der Liste bis der Bildschirm zu Ende ist. Sinnvoll wäre aber, wenn die bearbeiteten Dateien immer oben in der Liste angezeigt werden und dann nach unten rutschen. Die Liste auf 50 … 100 Einträger zu begrenzen wäre auch sinnvoll.

Danke für Eure Arbeit.

[Enhancement]　432Hz Batch Converter - show more decimal places when calculating the source pitch

As of at least v3.3, only a single decimal place is shown when calculating the source pitch, e.g. 440.3 or the like.

My suggestion is simple then—show more decimal places! Ideally I'd like to see at least 3 decimal places assuming that the value is being rounded to the nearest, otherwise 4 decimal places would be better it it's not rounding the displayed pitch value to the nearest (but even 2 decimal places would be much better than the current one decimal place).

　
The gist is that sometimes you need to know the percentage change in song length when using the "skip tempo adjustment for quality" setting, mainly for games and visual novels that have secondary metadata specifying which samples are where a given song is supposed to loop, and showing more decimal places for the source pitch lets you correctly calculate what the proper samples for looping and the like should be.

Additionally there are times where you can't even re-encode the source audio whether due to concerns over lossy--to-▶lossy transcoding or because it's some proprietary ADPCM format, in which case you next best option is to instead change the playback sample rate directly. This however requires you to know what the destination sample rate should be for a given piece of audio, and displaying more decimal points for the source pitch detection would allow you to accurately calculate what sample rate you should use.

　
Currently the only way to calculate the correct values for either scenario is to use a different program, such as foobar2000 or Audacity, and divide the total amount of samples your source has by the total amount of samples in the converted output and then take that total and multiply it by the original sample point for looping (1st scenario) or by the original sample rate (2nd scenario).

Constant bitrate

The program keeps converting mp3s to VBR and not to CBR despite the output settings.
Noticed even on the latest version (2.2)

Best regards
Misho Petrov

feature suggestion:

Maybe songs that are close to other good frequencies (such as 444hz) should get adjusted to 444hz instead of 432 hz.

fix

Log.txt
please fix this problem

Problems with statrtup

Every time I start the application it writes me the same error message: Unhandled exception........ The invocation of the constructor on type 'MediaPlayer.WindowsMediaPlayer'

I tried re-installing the program a number of times but it didn't fix anything....... It worked fine up until now but suddenly it kept showing this error message, that even re-installing doesn't solve...... It has nothing to do with Windows Media Player, because I never had it installed on my computer in the first place, and it did work for a while...... So I have no idea what's up with it. Here's the log file it keeps showing me:
Log.txt

ridicolous amounts of time to convert to 60fps

It's atrocious

Bug or intentinal? Sample rate setting is forgotten when closing the program

The format (WAV, FLAC, etc) setting and the bitdepth (24bit, 32bit, etc) settings are all remembered when you close the program.

But the sample rate setting is forgotten when you close the program and always resets back to "Source".

Is this a bug, or is this intentional behavior? The json file that contains the settings even has a field for "Sample Rate" but it doesn't seem to make a difference...

For reference, I'm using the x64 appimage on Linux Mint Xfce 20.3 that's been fully updated as of 2023-05-07.