From a quality perspective, what is better: turning volume up in the software, in the OS, or on the speakers?



If music isn't loud enough, how do I get the best quality (even if the difference is in fact so small it's negligible)?

  • By making the music louder in my music player, game or other sound-producing software program?
  • By raising the volume at the operating system level (for instance, by clicking the speaker icon in the Windows notification area and turning the volume up)?
  • By turning the volume up on the amplifier or speakers that are attached to your computer, and thus changing the volume on the hardware?

Does programs vs. OS matter? Does software vs. hardware matter?


Posted 2012-10-24T16:47:37.613

Reputation: 4 127


Generally you will want to avoid 100% on anything, but in particular on any analog controls. As your get close to the 100% you may run into audio clipping. I generally set my speakers volume to be ~60%, then adjust the computer until I get a comfortably loud sound. Then I always the speakers.

– Zoredache – 2012-10-25T07:16:44.537

2I get best sound by turning up the S/W volume to 99% then slowly up the volume on the speakers till perfect. The quality of the speakers counts for most of the quality. I am using Ubuntu 12.04 – peterretief – 2012-10-26T12:03:42.593

Example from my car AUX input: when i am turning the volume on my device to 100% and then regulate the volume on my radio the sound is not quite excellent. To make the quality better i turned the Volume on my device to 50% and then made it just louder in radio. – ekussberg – 2012-10-26T14:11:55.220

4@Zoredache Actually, 100% digital level is no problem at all unless there is some kind of audio processing going on in the signal path. In fact, most digital sound cards are set to a fix volume of 100% with no option to change it. – bastibe – 2012-10-26T15:19:10.087

Simple question gets a simple answer. Software 70-85% hardware can max out. Used with single or double amps. Loudness dependent in that order. – None – 2012-10-27T09:42:28.997



Program vs. OS generally doesn't matter. What matters is whether you're adjusting volume in software or in hardware.

Reducing volume in software is basically equivalent to reducing the bit depth. In digital audio, the signal is split up into distinct samples (taken thousands of times per second), and bit depth is the number of bits that are used to describe each sample. Attenuating a signal is done by multiplying each sample by a number less than one, with the result being that you're no longer using the full resolution to describe the audio, resulting in reduced dynamic range and signal-to-noise ratio. Specifically, every 6 dB of attenuation is equivalent to reducing the bit depth by one. If you started with, say, 16-bit audio (standard for audio CDs) and reduced the volume by 12 dB, you'd effectively be listening to 14-bit audio instead. Turn the volume down too much and quality will start to suffer noticeably.

Another issue is that these calculations will often result in rounding errors, due to the original value of the sample not being a multiple of the factor by which you're dividing the samples. This further degrades the audio quality by introducing what's basically quantisation noise. Again, this mostly happens at lower volume levels. Different programs might use slightly different algorithms for attenuating the signal and resolving those rounding errors, which means there might be some difference in the resulting audible signal between, say, an audio player and the OS, but that doesn't change the fact that in all cases you're still reducing bit depth and essentially wasting a portion of the bandwidth on transmitting zeroes instead of useful information.

This PDF has more information and some excellent illustrations if you're interested in learning more.

The result of reducing the volume in hardware depends on how the volume control is implemented. If it's digital, then the effect is much the same as reducing the volume in software, so there's probably little to no difference in which one you use, in terms of audio quality.

Ideally, you should output audio from your computer at full volume, so as to get the highest resolution (bit depth) possible, and then have an analogue volume control as one of the last things in front of the speakers. Assuming all the devices in your signal path are of more or less comparable quality (i.e. you're not pairing a cheap low-end amplifier with a high-end digital source and DAC), that should give the best audio quality.

@Joren posted a good question in the comments:

So if I want to set software volume control to max, how do I deal with my analog controls suddenly having a super tiny usable range? (Because even turning the analog volume to half is way too loud.)

This can be a problem when the volume control is part of an amplifier, which is probably the case with most computer setups. Since an amplifier's job is to, as the name suggests, amplify, this means that the volume control's gain ranges from 0 to more than 1 (often much more), and by the time you've turned the volume control to the halfway point, you're probably no longer attenuating, but actually amplifying the signal beyond the levels you set in software.

There's a couple of solutions to this:

  • Get a passive attenuator. Since it doesn't amplify the signal, its gain ranges from 0 to 1, which gives you a much larger usable range.

  • Have two analogue volume controls. If your power amplifier or speakers have a volume or input trim control, that will work great. Use that to set a master volume level so that your regular volume control's usable range is maximised.

  • If the previous two aren't possible or feasible, simply turn down the volume at the OS level, until you've reached the best compromise between the usable range on the analogue volume control and audio quality. Keep individual programs at 100% so as to avoid several bit depth reductions in a row. Hopefully there won't be a noticeable loss in audio quality. Or if there is, then I'd probably start looking at getting a new amplifier that doesn't have as sensitive inputs, or better yet, has a way to adjust input gain.

@Lyman Enders Knowles pointed out in the comments that the issue of bit depth reduction does not apply to modern operating systems. Specifically, starting with Vista, Windows automatically upsamples all audio streams to 32-bit floating point before doing any attenuation. This means that, however low you turn the volume, there should be no effective loss of resolution. Still, eventually the audio has to be downconverted (to 16-bit, or 24-bit if the DAC supports that), which will introduce some quantisation errors. Also, attenuating first and amplifying later will increase the noise floor, so the advice to keep software levels at 100% and attenuate in hardware, as close to the end of your audio chain as possible, still stands.


Posted 2012-10-24T16:47:37.613

Reputation: 21 756

TL;DR: Keep software levels at 100% and attenuate in hardware, as close to the end of your audio chain as possible – Edgar Griñant – 2017-07-19T12:42:01.247

32Some software allows increasing volume beyond "100%", or moving all the sliders in the equalizer to the top, what about that? It generally sounds a lot worse... – Daniel Beck – 2012-10-24T17:21:03.840

15@DanielBeck: In general, it's not recommended increasing volume above 100% unless you know the sound won't saturate (waveform won't clip, but that's hard to tell without a program to show the waveform, such as Audacity) or don't care about it (some sounds e.g. explosions and gunfire in games/movies when clipped don't actually sound worse to me). – Gnubie – 2012-10-24T17:35:33.947

4First, great answer Indrek. Spot on. But I should also mention that I've noticed even worse audio quality when I have several levels of volume sliders (the app itself, the system (software mixer) volume, and the hardware) and all of them are at less than max. volume. So if possible, start out with the application and start turning volumes up at the "lowest" level possible (closest to the application) and work your way "outward" turning volumes down on the higher level devices. So your volumes should go 100% (app) -> 100% or nearly 100% (operating system) -> much lower (for the amplifier). – allquixotic – 2012-10-24T17:37:45.327

5@DanielBeck That's basically just multiplying the quieter samples by a number greater than 1, or multiplying all samples by > 1 and then capping each one at the maximum bit depth. It generally sounds worse due to the reduced dynamic range, as well as clipping (distortion) introduced during the process. – Indrek – 2012-10-24T17:38:01.583

2RE: volume beyond 100%, that is impossible at a hardware level, so it basically means just upping the amplitude, which, if it isn't overdriving, is OK but won't give you the same quality as you'd experience if your source data were originally that loud. – allquixotic – 2012-10-24T17:38:33.967

@allquixotic Yes, that's exactly how it should be done - software volume controls maxed and analogue hardware control used for attenuation. – Indrek – 2012-10-24T17:39:38.070

1@Indrek, this is also why PulseAudio on Linux went to "flat volumes" where it compressed a number of volumes (stream volume, software mixer volume, ALSA software volume, and ALSA hardware volume) into a single volume, with the option to adjust app volumes relative to one another for the sake of being able to hear Skype over a movie (for example) but otherwise it does a minimum of attenuation in digital space. – allquixotic – 2012-10-24T17:41:29.613

So how come I sometimes get distortion if I set two volume controls really far apart? I haven't noticed this recently, was it just an implementation issue with some old software I was using? – Canageek – 2012-10-24T21:52:53.517

@Canageek I assume you're talking about software controls? If you set the first one very low, the bit depth would be greatly reduced, which would introduce a lot of noise and digital artifacts, which were then amplified by the second control and could be perceived as distortion. That's just a guess, though, without knowing the specifics of the setup. – Indrek – 2012-10-24T21:58:09.763


Concerning @DanielBeck's comment, even for silent signals that don't cause cutoffs it may be a better idea to use normalization instead. That is however mostly only available for playback of sound files known in advance, so Skyping with someone who didn't configure their mic right will still require that 150% setting...

– Tobias Kienzler – 2012-10-25T06:06:58.323

@ooo even in software: BBC iPlayer has 0-11 for the volume controls. – Richard Gadsden – 2012-10-25T09:14:51.340

2So if I want to set software volume control to max, how do I deal with my analog controls suddenly having a super tiny usable range? (Because even turning the analog volume to half is way too loud.) – Joren – 2012-10-25T20:40:20.557

"Reducing volume in software is basically equivalent to reducing the bit depth" - and reducing bit depth reduces volume range, not "quality" (by which you mean waveform fidelity). If you're reducing volume at the same time you're reducing volume range, there should be no loss in fidelity because of that. – None – 2012-10-26T08:37:43.180

@JoeWreschnig Actually, by "quality" I mean more than waveform fidelity (which does suffer too) - reducing bit depth also reduces dynamic range and increases the noise floor, which contribute to the overall reduction in audio quality. – Indrek – 2012-10-26T09:20:35.110


Modern PC operating systems (Windows Vista and newer, OSX) up-convert all audio to 32-bit floating point before doing volume adjustments. It's no longer true that using the software volume control destroys resolution or effective bit depth. More info here:

– Lyman Enders Knowles – 2012-10-26T16:22:42.090

1I think it's worth noting that this explanation, while excellent, may not apply to the situation where you've got external speakers plugged into a computer's headphone jack, which for most laptops is the only output jack there is. (E.g., plugging speakers into the jack on the side of a MacBook.) In that case, your system volume is acting as the "amplifier knob" versus a signal attenuator. For this I am going to guess (not sure why exactly) that maximum output volume from the headphone jack will not sound the best on the speakers. – Yetanotherjosh – 2012-10-26T21:15:56.100

@JSW yes I agree - most people will actually be doing the digital to analog conversion on a PC sound card, so surely turning that upto max is effectively amplifying the analog signal. The answer should be edited to differentiate between DAC done inside the PC sound card and external DAC. – rgvcorley – 2012-10-27T00:15:19.060

@rgvcorley Regardless of whether the DAC is external or built into the PC sound card, the OS and application volume controls are still in the digital realm and happen before the digital-to-analogue conversion. The question is about those software controls vs. an analogue control, like on an amplifier or speaker(s). – Indrek – 2012-10-27T08:26:05.700

1A couple points not particularly mentioned: (1) A cheap delta-sigma DAC may have 16- or even 24 bits of resolution, but will generate noise proportional to signal strength, such that it may only have 12 bits of useful resolution when producing a full-scale signal. Scaling the signal down by 20dB (a factor of ten in voltage--a pretty big reduction in loudness) would still leave 12 bits of useful resolution. Because of this, there's not much advantage to trying to maximize one's use of the DAC range; (2) audio processing, both analog and digital, will often cause a signal to go beyond... – supercat – 2012-10-27T17:24:03.123

...its envelope. For example, if one takes a square wave with an amplitude of +/- 1.0 volt, and puts it through a low-pass filter that removes high-frequency components while leaving lower ones untouched, the resulting signal will have a amplitude peaks that go well beyond +/- 1.0 volts. If one were to pass two +/- 0.5 volt square waves of 2000 and 3001Hz through a 10Khz low-pass filter, on equipment which clipped at 1.0 volts, the 1.0-volt clipping would likely cause noticeable distortion which would sound very different from if the amplitudes were cut by half. – supercat – 2012-10-27T17:31:10.513

1If software volume controls work by reducing the bit depth, why does my music still sound OK even when I have iTunes' volume set to around 5% and use a headphone amp? According to your answer, I would expect iTunes to reduce the bit depth to around 1 or 2 bits, which would sound like garbage noise. But it sounds fine (at least with a casual comparison). – Hank – 2012-10-29T16:39:49.513

"OS and application volume controls are still in the digital realm and happen before the digital-to-analogue conversion" -- For application volume settings I'd agree, but the OS volume interfaces to the sound card hardware which controls the post-DAC gain on the output amplifier. @rgvcorley the issue is not internal vs external DAC, it's whether volume control is applied pre-DAC or post-DAC. Attenuation pre-DAC loses bit depth, post-DAC just makes it quieter. I stick to my point that driving external speakers from a headphone output is an important counterexample to the answer. – Yetanotherjosh – 2012-10-29T17:59:28.747

@JSW Do you have a source for the claim that OS volume control affects post-DAC gain on the sound card? Also, I'm not sure what your point is about driving external speakers from a headphone output. Why is that an important scenario? The OS volume control doesn't act differently than in other cases, and more often than not you still have an analogue volume knob available on the speakers themselves. – Indrek – 2012-10-29T18:04:57.273

@indrek, I don't think either of us have sources for sound card signal flow diagrams, and it would obviously depend on the soundcard. But just consider this: it would simply be bad design for the gain of a headphone output amplifier to be constantly maximized and have music attenuated pre-DAC, for exactly the reasons of bit-depth loss you've explained, not to mention the noise floor problem: maximum amping of the noise floor for a signal that's making use of only a small fraction of the headroom. It just doesn't make sense. Some soundcards might do it that way, but it'd be crappy. – Yetanotherjosh – 2012-10-29T18:20:50.497

@JSW Actually, bit depth loss would appear not to be an issue with modern operating systems which upsample to 32-bit floating point before summing and attenuating, per a comment above. That contradicts your claim, however, because it means OS master volume has to be digital and pre-DAC. – Indrek – 2012-10-29T20:14:12.783

What would you tell about USB DACs which have special utility to control volume even below OS-level? (for example, I own "ESI Dr.DAC nano") I always thought that this thing (almost) directly controls the amplifier powers (analog scaling) instead of doing digital scaling. Am I wrong, or not? – Display Name – 2014-02-17T08:42:47.763


Basically, in sound, the closer to the physical source the better, to have a clear signal. Each physical stage will add noise. Earlier, stronger.

When a signal is amplified, any noise in the signal will also be amplified. A stronger signal means less noise compared to the signal. Therefore as it gets passed down the chain there will be less noise.


Posted 2012-10-24T16:47:37.613

Reputation: 7 993

11Quite true in the general case, but Qqwy's question asks more about software vs hardware volume control. – Gnubie – 2012-10-24T17:38:25.537

2So does this mean speaker or software? – Peter Ajtai – 2012-10-26T19:02:07.680

But sometimes early stages can suffer from increased nonlinear distortions when amplifying too much. And this is worse than just random noise. – Display Name – 2014-02-17T08:44:18.087


Typically, I like to have my software levels and OS levels as loud as possible. Since these sources are generally not amplified, their decibel ceiling should be 0dB; Essentially, they can't clip.

I then make sure that this sound goes directly to a single amplified destination, such as digital headphones (via USB), speakers with a volume knob and power supply, or an amp. I try to avoid chaining amplified devices because they can start to overdrive each other and cause clipping. Even individually, the amplification can result in clipping if the volume is turned up too high.

Since these can clip, I tend to keep these sources around the 50% volume range since that's normally where they're comfortable. It also affords the flexibility of increasing or decreasing the volume if the software/OS level is lower than usual.


Posted 2012-10-24T16:47:37.613

Reputation: 312


This definitely depends on what hard and software you are using. I am using a computer connected by this audio cable with two 3.5 jack plugs with a receiver, and if I put the sound on low on my computer (software) and high on the receiver, I hear a lot of noise. This probably has to do with amplifying not just the sound but also the noise that is being picked up by different components. Whenever I do this, I also hear noise when I'm not playing music.

It's different with my laptop tho, this is connected to the same receiver with an optical S/PDIF cable (digital) here I can put my volume on 100% on the receiver (my neighbors hate this!) It is really really loud and I can just turn the volume down on my laptop without any noticeable loss in sound quality. I do this because I have volume buttons on my keyboard and the receiver is quite far away.

Steven Stip

Posted 2012-10-24T16:47:37.613

Reputation: 231

+1, the accepted answer doesn't mention this, but this kind of amplified noise is, in my opinion, much more noticeable than the reduced resolution caused by reducing the volume in software. The amount of noise is very dependent on the quality of your hardware though. – yngvedh – 2012-10-26T14:34:50.660

And of course the technique to transfer the audio! Quality sometimes just doesn't matter(in hdmi cables for example) – Steven Stip – 2012-10-29T22:35:45.317


One error which I continue to see is end-users adjusting the volume via the particular program in use, only to later increase or decrease the volume via the soundcard (OS mixer, if you will).

Obviously, this creates confusion and does not allow for a predictable level of volume when launching other programs.

A simple solution--and the one I've employed for numerous years--is to establish a base level at both the hardware and OS level. By setting a permanent volume level in hardware and a permanent output level in software, you establish a standard to which you can compare the output of any program you use, adjusting the volume IN the specific program as desired (the advantage being that you will know what level volume you will receive from the specific program in the future).

Of course, to derive optimal benefit from both your amplifier and soundcard (OS), you must first set the volume of your amplifier to the maximum level afforded by the topology, but below unacceptable or undesirable levels of distortion. (Unfortunately, many low-powered 'class-D' audio amplifiers perform acceptably to a degree, but anything beyond that point [often, anything beyond 33 or 50 percent beyond its rated maximum output], often results in audible levels of distortion [as well as compression of dynamics and other undesirable effetc]. If you happen to have an audio amplifier with very low distortion at its maximum rating [provided the rating is of a weighted standard and not useless, like unweighted and measured only at 1kHz], you may have the liberty to set the output of your audio amp at maximum [under clipping range, of course; 'maximum' being contingent on the voltage of the input'. I remember being able to do this with amplifiers from Denon, Adcom, Hafler, and Nikon, in times past.)

The output of audio circuitry in some motherboards leave a lot to desire. In dedicated soundcards, the selection of high-quality soundcards is limited. For integrated audio circuitry, I advise selecting a volume level of no more than 2/3rds of the total range--and leaving it at that volume. (I know that is not scientific in its method, but from testing integrated outputs in many motherboards, I've noticed that distortion and other undesirable effects increase considerably as the output of the circuit approaches its maximum. Limiting the 'OS' level to 2/3rds (or 66%, or for the benefit of brevity and an easy to remember number, 70 [on a scale of 1 to 100; closer to 66% would be 66 on a scale of 1 to 100]) has served me well (while foregoing the need to perform exhaustive tests).

P.S. For the benefit of the initiated (or obsessive-compulsive)--and before an audiophile or engineer goes on a diatribe--I am well aware of the fact that setting the slider at 2/3rds level or the approximate 66 on a scale of 1 to 100 does NOT represent an actual output level of 66% of the total [the actual output will be lower], but it is a quick approach to obtaining an approximation of the cleanest output available from a motherboard's integral audio circuitry. P.P.S. The information provided assumes analog circuitry. If you are using digital circuity (SPDIF, Optical, other similar), you may set the soundcard ('OS') level to maximum with little risk of noticing a difference in the quality of the output from the audio circuity.


Posted 2012-10-24T16:47:37.613

Reputation: 111

2Having controls for each audio source as well as for overall volume is good, if one adopts the principle that one should adjust the volume of an audio-source if most of one's audio sources are playing at a reasonable volume but there's one that's too loud or too soft; one should adjust the overall volume if there's a change to how loud one wants sound to be (e.g. because someone is vacuuming or sleeping nearby). In most situations, one would probably want to change the overall volume more often than the individual-source volume, though ironically many programs facilitate the latter. – supercat – 2012-10-27T17:18:19.213


From a purely empirical standpoint, when I turn up my speakers all the way, I hear static.

I hear this static even if there is otherwise no sound coming out of my speakers.

So, I always max out the volume on the computer by maxing it in both the program and os, and then I try to keep the volume dial as low as possible on my speakers to minimize the static noise.

This might be a by product of my #@#!@%* speakers, but I assume many have speakers just like mine.

Peter Ajtai

Posted 2012-10-24T16:47:37.613

Reputation: 892

That static is literal noise, most of it internal to the electronics, but external RF signals are also picked up. I was once able to pick up a local radio station on my sub-woofer, with nothing but the power connected. – Baldrickk – 2018-04-09T15:11:46.160


I currently turn up the volume in software/OS to 100% and turn it down on the hardware side, but for a much simpler reason:

In my previous PC, the sound card generated noticeable white noise with a constant volume, no matter what i set the volume in the OS to. Regulating the sound on the hardware side helped reducing that noise.

Markus Unterwaditzer

Posted 2012-10-24T16:47:37.613

Reputation: 193


I would say hardware, but there's such thing as a standard volume for most applications.

However, DTS seems to be one of the exceptions based on the experience that if I play DTS movies I don't adjust the receiver volume level when I go from movie to movie and still find it comfortable.

If it is possible, I would rather have something that would output at the same level as DTS in order to remain comfortable.

That being said, for each OS they also have default system sounds. I would say you set your level against that volume level and let the OS deal with the volumes.

Archimedes Trajano

Posted 2012-10-24T16:47:37.613

Reputation: 879


When you send an audio signal through a chain of volume knobs (of any kind: analog, digital, physical, software), set each one as loud as you can without clipping or distorting. Otherwise, you needlessly decrease the signal's dynamic range (aka number of bits, aka quietness of hiss). Use only the last knob to adjust the volume of the signal exiting the chain. That optimizes quality, for anything from a laptop videogame to a live transatlantic symphony orchestra broadcast.

Camille Goudeseune

Posted 2012-10-24T16:47:37.613

Reputation: 1 361


This question is too varied, but if you must get an answer it depends on several situations...

  1. Hardware, is it internal speakers, external speakers, headset, etc?
  2. How loud are we talking? Level 1 to level 10? or just a nudge?

Hardware, it depends if you have good speakers to begin with, a nudge with the speakers either via software or hardware may be noticeable if they are external, and a specific name brand, but if they are cheap, you may need to increase more than a nudge.

As for software, adjusting the software is always preferred before adjusting the hardware if it's external, because software adjusting today is sometimes easier... and hardware although still easy, it depends if you have it connected to an external equalizer, or something all together different.

Some audiophiles will tell you that once you get the hardware you want to have it, you will never need to touch it again, except to adjust the volume... others will say that software adjustments are better...

Again, you leave out a lot of variables, and this really is too broad, and may want to re-adjust the question by putting more details into it.

Matt Ridge

Posted 2012-10-24T16:47:37.613

Reputation: 271