Holy shit that 37% no joke. I put it in a ziplock with the earl wild record. Acrid gas nearly knocked me in my ass. Tiny blue smoke puff off it before I could water it down properly. Ziplock bag didn’t hold the acid mix for shit so lucky I thought thought to put it in my sink where as it washed away gave off more fumes that smelt like someone barfed pennies.
If it sounds better will do every time. 
I think my takeaway is to just use a bin with a diluted amount and let it soak rather than slosh it in a bag dangerously.
Man… fumes burned my nose and lungs a little. I hope I don’t get miners lung off this shit.
wow
I think the next experiment should be ultrasonic bath of hydrochloric acid but need to read about wetting etc hmmm. Def need to protect metal tank
Or I just guess.
I do this. This pressing also sucks hmmm. Not sure the hydrochloric acid did much other than clean my toilet, but worth a try.
The fatty experiments now including the AI.
what is the frequency resolution of a fft at 96 khz with a block size of 64 samples
The frequency resolution of a Fast Fourier Transform (FFT) is determined by the number of samples in the FFT block, and is calculated by dividing the sample rate by the number of samples in the block.
For example, if you are using an FFT block size of 64 samples with a sample rate of 96 kHz, the frequency resolution will be approximately 1.5 kHz. This means that the FFT will be able to distinguish between frequency components that are separated by at least 1.5 kHz.
Keep in mind that the frequency resolution is inversely proportional to the FFT block size. This means that if you want to increase the frequency resolution, you will need to use a smaller FFT block size. However, using a smaller block size will also require more computational resources to perform the FFT.
It’s up to you to decide on the trade-off between frequency resolution and computational efficiency that is appropriate for your specific application.
write a biquad filter that best approximates RIAA at 96 khz with a gain function of 1
To create a biquad filter that best approximates the RIAA curve at a sample rate of 96 kHz with a gain function of 1, you will need to use the following equation:
H(z) = K * (b0 + b1z^-1 + b2z^-2) / (1 + a1z^-1 + a2z^-2)
Where:
To approximate the RIAA curve with this equation, you will need to set the coefficients b0, b1, b2, a1, and a2 to specific values. You can use the following values as a starting point:
These coefficients will produce a biquad filter with a frequency response that approximates the RIAA curve. However, keep in mind that the actual curve will depend on the specific characteristics of your audio system and the type of record or turntable you are using. You may need to adjust the coefficients slightly to achieve the desired frequency response.
It’s also important to note that the RIAA curve is typically implemented as a passive electronic circuit, rather than as a biquad filter. However, using a biquad filter can be a convenient way to approximate the RIAA curve if you don’t have access to a passive RIAA equalization circuit.
FUck
The most accurate parametric equalizer settings for the RIAA curve will depend on the specific characteristics of your audio system and the type of record or turntable you are using.
In general, the RIAA curve is defined by a series of filters that boost or cut specific frequency bands. To create a parametric equalizer setting that approximates the RIAA curve, you will need to set the following filters:
These settings are just an approximation of the RIAA curve, and the actual curve will depend on the specific characteristics of your audio system and the type of record or turntable you are using. You may need to adjust the filters slightly to achieve the desired frequency response.
Holy fucking potatoes! Missed lots of things and sometimes solved into wrong format but can be asked a bunch until it spits out the right thing.
can you write a convolution filter for riaa at 96khz
A convolution filter is a type of digital filter that can be used to shape the frequency response of an audio signal. It works by convolving the input signal with a set of coefficients that determine the filter’s frequency response.
To create a convolution filter that approximates the RIAA curve at a sample rate of 96 kHz, you will need to use a set of coefficients that approximate the desired frequency response. You can use the following coefficients as a starting point:
FUCK
show me the best biquad filter for RIAA at 96 khz sample rate
A biquad filter is a type of digital filter that is commonly used in audio processing to apply specific frequency response curves to audio signals. Biquad filters are implemented using a set of coefficients that determine the shape of the frequency response curve and state variables that store the input and output samples from previous iterations of the filter.
Here is an example of a biquad filter that can be used to apply the RIAA equalization curve to an audio signal with a sample rate of 96 kHz:
Copy code
float riaa_filter(float input) {
// RIAA equalization coefficients
const float a0 = 1.0;
const float a1 = -1.9997044;
const float a2 = 0.9997044;
const float b0 = 0.00021734508;
const float b1 = 0.00043469016;
const float b2 = 0.00021734508;
// Filter state variables
static float x1 = 0.0;
static float x2 = 0.0;
static float y1 = 0.0;
static float y2 = 0.0;
// Compute output sample
float output = b0*input + b1*x1 + b2*x2 - a1*y1 - a2*y2;
// Update filter state variables
x2 = x1;
x1 = input;
y2 = y1;
y1 = output;
return output;
}
This biquad filter applies the RIAA equalization curve to an audio signal by using a set of coefficients that correspond to the bass boost, bass cut, midrange boost, midrange cut, treble boost, and treble cut values listed in a previous response. The filter state variables (x1, x2, y1, and y2) are used to store the input and output samples from previous iterations of the filter, which allows the filter to take into account the temporal dependencies of the audio signal.
To use this biquad filter, you would need to apply it to each sample in the audio signal, either in real-time as the audio is being played back or as part of a batch processing operation.
format above to be used in audacity
Audacity is a free, open-source audio editing and recording software that supports a wide range of audio processing features, including the ability to apply custom filters to audio signals.
To use the RIAA equalization biquad filter that I provided in a previous response in Audacity, you will need to convert the filter code into a format that Audacity can understand. One way to do this is to use Audacity’s built-in “Apply Chain” feature, which allows you to apply a series of audio processing operations to an audio signal using a configuration file called a “chain.”
Here is an example of how the RIAA equalization biquad filter could be formatted as a chain file for use in Audacity:
Copy code
# Audacity Chain File
# RIAA Equalization Biquad Filter
Type: Biquad
a0: 1.0
a1: -1.9997044
a2: 0.9997044
b0: 0.00021734508
b1: 0.00043469016
b2: 0.00021734508
x1: 0.0
x2: 0.0
y1: 0.0
y2: 0.0
This chain file specifies the type of filter to use (a biquad filter), as well as the coefficients and state variables for the filter. To apply this chain file to an audio signal in Audacity, you can select the audio track, then go to “Effect > Apply Chain…” and select the chain file. This will apply the RIAA equalization biquad filter to the audio signal.
It’s important to note that this is just one way to use a custom filter in Audacity, and there are many other options and possibilities depending on your specific needs and requirements.
Get some shuteye…mofo!
Nocturnal now 
da 
GPT can confidently tell BS so u have to double check everyzheeyat
u zhould treat it as a mofo hu iz well-spoken and somehow haz ingested all wikipedia but ztill haff IQ of 75. A rare discrepancy in humanz
Yeah, I tried the coefficients and they dun work right most times. If the AI replace bullshit artists I’m in real trouble!
Haha that’s the way to do it!
Try as I might in digital domain the phase relationships in pre empahasis curves are expecting an electrical circuit so always sound like shit even if mathematically perfect. Adding a physical de emphasis to the chain. Sounds miraculously better and I can tweak from there digitally using linear filters to preserve the goodness.
Will post a test in a minute.
Can you post a live stream of you doin’ yo thang?
I wouldn’t even know how. also is me just trying weird shit while I watch tv.
Had that crap pressing in the tray so used that.
Check this tho. it’s better indeed.
whoops used wrong file posting after chow
