What's the consensus on adjusting the volume in OS vs the speakers?

What's the consensus on adjusting the volume in OS vs the speakers?

Speakers?

system volume and everything digital goes to 100% to use the full range of the DA
then adjust in the analog domain

9/10 times I would rather adjust via a physical knob, unless I'm watching porn then its scroll wheel adjusting

This

90% for the convenience of being able to use my keyboard if necessary
When I actually work with music i use ASIO obviously, which removes microsofts volume control among other unecessary processing so "100%" in that case

I keep system volume at 50% so that if necessary I can adjust that, but most of the time I use this knob.

/thread

>implying a basic resistive adjustment dulling a digital signal is better than just sending a smaller precise digital signal
Do you even know where your gain is versus your cut?

He doesnt have one of these

Assholes use speakers

I turn my speakers to max and adjust volume with my keyboard volume wheel on my l33t G.Skill gaymen keyboard.

I agree with this.

All of my hardware is old and busted. I try to not fiddle with the analog knobs and wires too much. If I disturb the equipment too much I'll get static or lose sound in one of the speakers.

So yeah, the analog stuff is fixed. And I adjust the main volume (and the volume in mpv if needed).

digital volume at 100, 50, or 0
use analog volume controls otherwise

That's a bit harsh, I'm sure there are people who use speakers that aren't assholes.

I can definitely recommend a nice set of studio monitors. Not to expensive and less ear harming than headphones.

Depends on your system. I run mine through a DAC and an amplifier and adjust the volume on the amplifier, keeping the OS sound around 60-70% and speakers set at 75%.

Generally not a good idea. Normally should leave several decibels of margin on top, usually 3-5 dB.
How much is 3 to 5 dB on your volume control? I don't know.

For convenience, I don't max either.

care to explain why?

speakers at 50% always and adjust with os volume control for exact and familiar increments.

that way you're throwing away half the SNR of your DAC, neat.

Speaker close to max bc and control everything from software. You also won't wear the buttons out.

Leaving some headroom helps to prevent clipping, which is far more noticeable than noise.

to reduce digital clipping

if you can adjust the gain of your sound cards DAC, there's typically a sweet spot where the dynamic range is better too, sometimes the digital volume directly sets the gain, depends on the driver

with typical SNR's over 100db who cares?

digital clipping is introduced over 0dBFS
which is at 100% system volume

yfw the speaker buttons changes the OS volume

the signal can still go over due to resampling

if you're using amplified speakers, you should be using line-out, not speaker-out

yeah, intersample peaks can when you resample. at frequencies where it's inaudible in 99% of all cases
if you're really anal about that then you can calculate how high a signal can go over for each possible samplerate conversion and it's max +1.5dB, so your 5dB headroom is still nonsense

the digital volume isn't logarithmic so at 50% you've still only gone down a few dB

depends on what arbitrary scale your os is using and the driver
I've absolutely seen it go down like 48dB at 50%

if you went down 48dB most common sounds would probably be barely audible, I don't believe any digital volume control goes virtually inaudible at 50%

>people exist who don't have their volume at 100%, then use the knob on the speaker/interface to adjust

I had no idea retards of this level existed.

absolutely
one example I carry around in my pocket is the ESI Dr. DAC with windows 7 and there's others.

Keep system vol at 100% for more effective and satisfying knob jogging

Set volume on reciever to just this side of pissing off neighbors.
Adjust volume digitally.

I started using the digital volume adjustment when my analogue potentiometer started screwing things up.
If you are using analogue output from your soundcard you should always use the analogue control from the speakers, otherwise you're amplifying your pc's noise at maximum level.

Pic related. It's my crappy old surround set

an external DAC doesn't really count

I have some crudely made speakers made from an old plasma tv. For some reason they chirped a lot at system vol 100%, so I kept it at 80% and then adjust the analog volume knob and the chirping dissapeared.

sounds like your converter has a problem

Waveform may exceed the values of the 0 dBFS samples, depending on the sample amplitude of the signals and how it is interpolated. The issue is more likely to occur with greater severity using audio that has high amplitude content near fs/2.

When you sample a signal, you quantize values that represent a continuous waveform in the time domain.
When you reconstruct a signal, you basically convolve (or this case multiply) the impulse response with each sample point. The "impulse" is scaled based on the value of each sample, and the combined values in continuous time from the "impulses" will assemble a waveform.

Consider this example:
>Assume a sequence of evenly spaced samples that are zero valued everywhere except for two adjacent samples. These samples (let's call them "N" and "N+1") are both the positive value full scale value. We'll use 1 and -1 for brevity, but any number of levels can be added.
>We will assume the system's impulse response is the sinc function. Even though a true sinc response is not realistic at all, it plays nice with the sampling theorem and it will make an interesting point [if I can get to it].

The midpoint between N and N+1 will take the value 4/π, or approximately 1.273. The reproduced waveform will take a value that many times greater than the nominal full scale value, or clip. In the digital domain, time shift the signal less than a full sample or resample the signal will result in digital over.


Virtually all current audio DACs, save for a handful of esoteric audiophile novelties and the pre-1990 gear, will oversample by default. Oversampling interpolates sample points between those originally given by the Nyquist rate signal.

Digital peak meters have traditionally been based on sample peak level, something that can undervalue the reconstructed peak by a significant amount. Peak limiter compression has been one of the staple tools of audio production, and intersample peaks can be regularly observed with any modern music

I set the alsa volume to 0dB (on a scale of +6dB to -128dB) on all the channels I'm likely to ever use, I set the monitor and headphone knobs to about 75%, and then I use the pulseaudio volume to make adjustments from there. Works great, except sometimes ALSA randomly sets both of the ears of the headphones to PCM 2 when I restart.

I don't know, but after getting a DAC(having volume control on the headphones as well) I've put everything on turbo and control the volume through the receiver or the DAC.

my speakers (with volume knob) are 10 feet away from me so i use OS volume control
i could get buy unpowered speakers and put the amp right next to me but i cbf

I never get this practice.

If I set my digital volume to 100% and rely on the analogue knob then I would most certainly destroy my eardrums and my neighbours' with the slightest of knob movement.

>the slightest of knob movement

Oh my God who else did this as well

British boarding school, with all that traditional school uniform, you cut a whole in one of your pocket so your friends can wank you off during morning assembly and prayer.

user...

get help

hahaah. the 'analogue knob'.

How do you deal with the cum afterwards?

What?

I'm fine, user.

:)

If you do it whilst still wearing your underwear it's easy to just change into a new pair when you get back to house before the first lesson starts.

...

...

these are garbage

For everyday use: system volume at 40% and speakers volume at 50%.

For audiophile use: system volume at 100% and adjust the volume with my headphone amp.

Sample overs can be induced by lossy coding, which resamples the signal in question.
Filter calculation may not respond intuitively given a large overload.
Now, how much all this affects real world audibility is not well determined. It does affect peak readings, and the reproducibility of measured values, such as a steady state sine that varies more than it ought to.

A "True Peak" meter has been proposed some years ago, for instance as one part of ITU Rec BS.1770. The design oversamples the input signal and interpolates the waveform with a filter to better detect intersample overs.


Let's reconsider the previous sequence with N and N+1 and the sinc interpolation.
This time, we'll make the two samples adjacent to those, N-1 and N+2, take the value of -1. We can have N-2 and N+3 be +1, and repeat the process as many times as we like.

... -1 +1 -1 +1 +1 -1 +1 -1...

(N-3) (N-2) (N-1) N (N+1) (N+2) (N+3) (N+4)


Again, we'll take the midpoint between N and N+1. Its value as it is related to the pairs can be expressed as a summation.
>(2/π) * Σ 1/(0.5+i)
The term "i" is to represent the contribution from each pair.

For an infinite number of samples ordered as such, the series is divergent to infinity.
In other words, we have an intersample over of infinite amplitude.

Now, about that example:
The sequence of sample values would be considered pathological. We also don't have an infinite number of samples to work with.
The sinc filter oscillation is slow to decay, and the given (pathological) sequence takes the most possible advantage of its oscillation. We can't really implement a sinc response either. We can have at best a truncated or tapered version in finite time, so a more appropriate calculation would involve finite terms, not an infinite series.
It does show that you can create a very large peak that the sample values can completely miss.

>What's the consensus
Literally fucking kill yourself memester.

It seems that I will have to teach you how to use a speaker in 7 simple steps

>wait until ~01 or any time that you know all of yoir faggot neighbors in your apartment block are asleep
>connect your overpriced cumpad to your loudest system you have, alternatively if you have a Toshiba TV connect to that as they are loud af
>maximum sound on speakers/TV
>maximum sound on cumpad
>use the VLC equalizer to make a slight, small but noticable treble boost
>use the compressor on low attack and release and decay, also add some dB to the bass here
>play eurobeat

I already got 2 warnings this way. After the third I will finally get my eviction notice (and my keks)

I usually up the volume on the speakers up to 30% and use my keyboard to up and down the volume as needed.

I have an on-board sound card, and with the volume up a 100% on the volume knob results in a hiss which is terrible, and i won't even be able to fall asleep.