5 ms·

I was surprised by the best answer as it seems to me to be more or less wrong. Let's ignore the discussion about dynamic range and bit depth etc., and assume t

I was surprised by the best answer as it seems to me to be more or less wrong. Let's ignore the discussion about dynamic range and bit depth etc., and assume that the volume control on your operating system controls the DAC rather than doing the stupid thing of digital volume reduction. The fundamental issue is signal to noise ratio on the analog line. If you turn the volume too far down on the computer and turn the volume up on your speakers, the sound on the analog line is too low with regard to the electrical noise and will be hissy. If you turn the volume up too much on the computer and turn the volume down on your speakers, then the signal will be so loud as to produce distortion either in the DAC or on the line itself. You're looking for a middle ground: as loud an output from the computer that you can produce without causing distortion in your loudest music parts. Once you've got that set, change the volume on the speakers to compensate.

deleted 14y ago
[deleted]
matheusalmeida 14y ago
I think you're absolutely right and it's not the first time and I'm sure it will not be the last time that a very verbose post (read well structured with references and mentioning technical terms) gets much more credit than it deserves. It would be nice to have some statistics about the number of lines of a reply on SO or HN and the amount of 'upvotes'. I think a lot of people would be surprised...
ZoFreX 14y ago
> assume that the volume control on your operating system controls the DAC rather than doing the stupid thing of digital volume reduction Is this a safe assumption though? As much as I consider myself knowledgeable in this area, this is way off my radar. Even if this assumption does hold, the answer does make a good point when it comes to per-application volume controls, which are certainly performed digitally. Your point on SNR is a very good one, particularly on the low-end, but I would question whether any computer at max volume would produce a signal strong enough to cause distortion. I generally run my computer at around 90% volume but that's merely a habit carried over from having an analog hifi stack.
- klodolph 14y ago
  It would be easy enough to test -- download Right Mark Audio (free, but Windows only), connect your audio input to output with a patch cable, and test at different volume levels.
- SeanLuke 14y ago
  > Is this a safe assumption though? No it's not -- I was just pointing out that he's going on about various stuff without hitting the critical meat of the issue at all.
- hammock 14y ago
  Guy with some sound knowledge here. It doesn't matter if the attenutation is done digitally (as described by the top SE comment) or in analog fashion. The answer is always the same. In the music biz the general rule is: you want to put as much gain as early in the chain as possible. In other words, start at the source of the sound, and working your way towards the speakers, turn it up as loud as you can without distorting it. Take the example of a guitarist. He turns his guitar up as high as he can, then the amp gain as high without distorting it. Then at the sound console. etc The reason for this the later in the chain you are, the more electronics and cable is involved which adds noise to your signal. Highest signal-to-noise is achieved by ensuring you have as much signal as you can running thru from the get-go. In the case of a computer, again it doesn't matter how volume changes are done- crank up your software as high as you need (but not beyond the point of distortion), then adjust it further at the speakers.
  - anthonyb 14y ago
    > Take the example of a guitarist. He turns his guitar up as high as he can, then the amp gain as high without distorting it. Then at the sound console. etc Without distorting it? Are you sure you have some sound knowledge? That doesn't sound like you know many guitarists... ;) qv. https://www.youtube.com/watch?v=4gDsbOraiqg https://www.youtube.com/watch?v=4gDsbOraiqg
    - iamdave 14y ago
mikeash 14y ago
Is it really possible to turn the volume up that high on a computer you're likely to encounter? I've yet to encounter a computer-like device that produces any noticeable distortion on the output when turned up all the way, if the other end has a volume control to make the final output happen at a reasonable level.
- klodolph 14y ago
  Lots of computers I've used allow audio gain to be set above unity, with noticeable distortion as a result. Next time you're in alsamixer on Linux, look at the fourth line down on the top which says "dB gain". It's not uncommon for it to go all the way up to +6 dB or +12 dB, which will almost certainly distort with common source material. This happens both on my desktop Linux system and my Linux laptop.
- cnvogel 14y ago
  Most consumer-hardware I know will have positive gain on the computer output (all built-in audio for my devices, the PCI cards with Ensoniq 1370 or 1371 I used earlier). On the other hand, a lot of the professional audio equipment I know have mixer-applications that has either 0dB as a maximum setting, or clearly labeled 0dB marks/buttons on the mixer application). You can verify it easily by playing either a 0dB sine-wave mp3 (first google hit: http://www.dr-lex.be/software/testsounds.html http://www.dr-lex.be/software/testsounds.html) or trying to get a sine-wave test-tone generator. You'll hear the point where distortion kicks in pretty drastic as soon as you reach the clipping level. With a spectrum-analyzer (e.g. FrequenSee on Android) even slight clipping will be very visible (peaks at 3x, 5x, 7x, ... the base frequency will suddenly appear).
mbell 14y ago
> assume that the volume control on your operating system controls the DAC rather than doing the stupid thing of digital volume reduction. What would be the difference? I've never seen a DAC with an analog volume control. Any DAC I've ever seen with volume control does so digitally, usually with toggable steps (+6DB, +12DB, etc) or rarely with a programmable gain. Either way the DAC is adjusting the incoming digital signal prior to conversion. It would be really strange to try to make an analog adjustment to the output before the reconstruction filter.
- cnvogel 14y ago
  First hit for a AC97 codec: AC ’97 SoundMAX® Codec AD1981B has an analog mixer stage between ADC/DAC and analog input/outputs. http://www.xilinx.com/products/boards/ml505/datasheets/87560554AD1981B_c.pdf http://www.xilinx.com/products/boards/ml505/datasheets/87560...
  - mbell 14y ago
    That is a 10 year old codec, not a DAC, a DAC is only one part of a codec.
    - makomk 14y ago
      Computers generally tend to have full-blown codec chips with analog mixers in them rather than standalone DACs, though.
      - mbell 14y ago
        Depends on the model, cirrus codecs used in almost all mac's do all their mixing digitally for example.
tiziano88 14y ago
A DAC is unlikely to generate additional distortion on its own only because its volume is too high, as it is (usually) designed to obtain a unit gain corresponding to the maximum level of digital signal at its input. If a 16-bit DAC maxes out its analog output when the input signal is - say - 2^14, then it means that it is effectively operating on a dynamic range of 14-bit, and should be advertised as such. Of course things are not so straightforward, as even the best DACs are not perfectly linear, but they definitely should not clip the output signal.
mbenjaminsmith 14y ago
Little late to the party here but you're wrong. Unless something is being done to the signal (eq, other processing, etc) then you want the digital side to remain at max volume (i.e., no reduction in bit depth). The audio will have been mastered close to 0 db with little or no clipping (exceeding the maximum amplitude possible) and should not be "too loud" for your amplifier. If it is there is something wrong with the gain configuration of your setup. As discussed, lowering the volume in software (which is what you likely mean by "controlling the DAC") is accomplished by reducing bit depth. By definition you're reducing the amount of signal present which makes your S/N worse. If by "controlling the DAC" you mean reducing the signal at the DAC's analog stage then there's nothing theoretically wrong with that (except that it's rarely if ever the case in computer sound cards or hi-fi DACs) as it is the equivalent to adjusting the volume on your (integrated) amp. Probably the most common exception to the above is people using a digital EQ to attempt to improve the sound of a mediocre computer audio setup. Most digital EQs allow for both "boost" and "cut" -- increasing or decreasing the amplitude of individual frequency bands. If you EQ the music the proper way you should only cut frequency bands in order to emphasize the other bands to suit your listening tastes. For example if you want to try to increase the amount of bass you should cut the mids and the highs and turn up the volume (on your amp) to compensate. EQing this way ensures that you're not forcing the signal into clipping by boosting too much. You still won't get great results because if you're lacking bass or highs then it's most likely because you're not running full range speakers. If the sound does change when you EQ it's most likely due to increased distortion, not because you're appreciably increasing the amount of bass present. That's not true for full range speakers but if you have full range speakers you probably wouldn't need to EQ in the first place. (Full range speaker are usually 20 - 50hz to 20khz.) If it's still not clear than think of it this way: Standalone CD players don't have volume controls. Their output will be "max volume" of what's on the CD. You can think of your computer and sound card like a (crappy) CD player. Make sure it's at max volume and adjust the level on your external amp or amplified speakers.
- asynchronous13 14y ago
  'As discussed, lowering the volume in software ... is accomplished by reducing bit depth.' No, reducing the amplitude of a digital signal does not change its bit depth. If the amplitude of the digital signal is reduced, then a couple of bits on the MSB side will be constant. However, reducing the bit depth would make bits on the LSB side constant. To state it differently, a lower bit depth increases the quantization levels and results in lower quality audio. Reducing the amplitude retains the same quantization level, and therefore has no reduction in audio quality (At least, not due to bit depth, it does change the SNR as you suggested)
JagMicker 14y ago
I've seen a lot of comments that seem to confuse digital and analog audio. Digital audio levels are fixed -- i.e. 0 dBFS (dB relative to digital "full scale"), thus you want the audio as close to full scale (0dBFS) as possible. This is the reasoning behind what you've probably heard described as the "loudness war". 0dBFS is the reference point beyond which no signal can exceed. Contrast that with analog audio, be it the DAC (digital-to-analog converter) on your soundcard that splits to a "line out" (likely 3.5mm a/k/a/ 1/8" TRS connector outputting audio at -10 dBV, the "consumer" level) and "headphone jack" (i.e. line level signal -> headphone amplifier). Once the audio signal becomes analog, you must then be concerned about gain staging. If you have a crappy sound card, you shouldn't reduce the analog output, because, in order to achieve the same loudness, you will have to increase the gain on your 'speakers' (a/k/a/ "pre-amplifier"; often integrated into cheap 'all in one' computer audio playback systems). Remember that the noise floor remains the same regardless of gain settings, so by reducing the "line out" level, you are only increasing the noise floor (i.e. level of noise). For all of you geeks out there... If you are looking to cut through the bullst and get a quality playback system, look into a setup that has a quality DAC and/with monitor controller, a separate preamp, and quality monitors. Benchmark has a great USB DAC, and or Lavry Engineering is what the professional mastering studios use (their DAC costs well over $20k, and not because of crap like 'voodoo diamond dust covered gold-plated platinum wires', but, in part, because it has an internal oven and temperature monitoring circuitry to keep the internals at precise temperatures). You can get a great monitor controller from Dangerous Audio or Coleman. Bryston preamp's are very popular in the professional setting, as are PMC and Bowers & Wilkins monitors.

deleted 14y ago

[deleted]

matheusalmeida 14y ago

I think you're absolutely right and it's not the first time and I'm sure it will not be the last time that a very verbose post (read well structured with references and mentioning technical terms) gets much more credit than it deserves. It would be nice to have some statistics about the number of lines of a reply on SO or HN and the amount of 'upvotes'. I think a lot of people would be surprised...

ZoFreX 14y ago

> assume that the volume control on your operating system controls the DAC rather than doing the stupid thing of digital volume reduction Is this a safe assumption though? As much as I consider myself knowledgeable in this area, this is way off my radar. Even if this assumption does hold, the answer does make a good point when it comes to per-application volume controls, which are certainly performed digitally. Your point on SNR is a very good one, particularly on the low-end, but I would question whether any computer at max volume would produce a signal strong enough to cause distortion. I generally run my computer at around 90% volume but that's merely a habit carried over from having an analog hifi stack.

klodolph 14y ago
It would be easy enough to test -- download Right Mark Audio (free, but Windows only), connect your audio input to output with a patch cable, and test at different volume levels.
SeanLuke 14y ago
> Is this a safe assumption though? No it's not -- I was just pointing out that he's going on about various stuff without hitting the critical meat of the issue at all.
hammock 14y ago
Guy with some sound knowledge here. It doesn't matter if the attenutation is done digitally (as described by the top SE comment) or in analog fashion. The answer is always the same. In the music biz the general rule is: you want to put as much gain as early in the chain as possible. In other words, start at the source of the sound, and working your way towards the speakers, turn it up as loud as you can without distorting it. Take the example of a guitarist. He turns his guitar up as high as he can, then the amp gain as high without distorting it. Then at the sound console. etc The reason for this the later in the chain you are, the more electronics and cable is involved which adds noise to your signal. Highest signal-to-noise is achieved by ensuring you have as much signal as you can running thru from the get-go. In the case of a computer, again it doesn't matter how volume changes are done- crank up your software as high as you need (but not beyond the point of distortion), then adjust it further at the speakers.
- anthonyb 14y ago
  > Take the example of a guitarist. He turns his guitar up as high as he can, then the amp gain as high without distorting it. Then at the sound console. etc Without distorting it? Are you sure you have some sound knowledge? That doesn't sound like you know many guitarists... ;) qv. https://www.youtube.com/watch?v=4gDsbOraiqg https://www.youtube.com/watch?v=4gDsbOraiqg
  - iamdave 14y ago

mikeash 14y ago

Is it really possible to turn the volume up that high on a computer you're likely to encounter? I've yet to encounter a computer-like device that produces any noticeable distortion on the output when turned up all the way, if the other end has a volume control to make the final output happen at a reasonable level.

klodolph 14y ago
Lots of computers I've used allow audio gain to be set above unity, with noticeable distortion as a result. Next time you're in alsamixer on Linux, look at the fourth line down on the top which says "dB gain". It's not uncommon for it to go all the way up to +6 dB or +12 dB, which will almost certainly distort with common source material. This happens both on my desktop Linux system and my Linux laptop.
cnvogel 14y ago
Most consumer-hardware I know will have positive gain on the computer output (all built-in audio for my devices, the PCI cards with Ensoniq 1370 or 1371 I used earlier). On the other hand, a lot of the professional audio equipment I know have mixer-applications that has either 0dB as a maximum setting, or clearly labeled 0dB marks/buttons on the mixer application). You can verify it easily by playing either a 0dB sine-wave mp3 (first google hit: http://www.dr-lex.be/software/testsounds.html http://www.dr-lex.be/software/testsounds.html) or trying to get a sine-wave test-tone generator. You'll hear the point where distortion kicks in pretty drastic as soon as you reach the clipping level. With a spectrum-analyzer (e.g. FrequenSee on Android) even slight clipping will be very visible (peaks at 3x, 5x, 7x, ... the base frequency will suddenly appear).

mbell 14y ago

> assume that the volume control on your operating system controls the DAC rather than doing the stupid thing of digital volume reduction. What would be the difference? I've never seen a DAC with an analog volume control. Any DAC I've ever seen with volume control does so digitally, usually with toggable steps (+6DB, +12DB, etc) or rarely with a programmable gain. Either way the DAC is adjusting the incoming digital signal prior to conversion. It would be really strange to try to make an analog adjustment to the output before the reconstruction filter.

cnvogel 14y ago
First hit for a AC97 codec: AC ’97 SoundMAX® Codec AD1981B has an analog mixer stage between ADC/DAC and analog input/outputs. http://www.xilinx.com/products/boards/ml505/datasheets/87560554AD1981B_c.pdf http://www.xilinx.com/products/boards/ml505/datasheets/87560...
- mbell 14y ago
  That is a 10 year old codec, not a DAC, a DAC is only one part of a codec.
  - makomk 14y ago
    Computers generally tend to have full-blown codec chips with analog mixers in them rather than standalone DACs, though.
    - mbell 14y ago
      Depends on the model, cirrus codecs used in almost all mac's do all their mixing digitally for example.

tiziano88 14y ago

A DAC is unlikely to generate additional distortion on its own only because its volume is too high, as it is (usually) designed to obtain a unit gain corresponding to the maximum level of digital signal at its input. If a 16-bit DAC maxes out its analog output when the input signal is - say - 2^14, then it means that it is effectively operating on a dynamic range of 14-bit, and should be advertised as such. Of course things are not so straightforward, as even the best DACs are not perfectly linear, but they definitely should not clip the output signal.

mbenjaminsmith 14y ago

Little late to the party here but you're wrong. Unless something is being done to the signal (eq, other processing, etc) then you want the digital side to remain at max volume (i.e., no reduction in bit depth). The audio will have been mastered close to 0 db with little or no clipping (exceeding the maximum amplitude possible) and should not be "too loud" for your amplifier. If it is there is something wrong with the gain configuration of your setup. As discussed, lowering the volume in software (which is what you likely mean by "controlling the DAC") is accomplished by reducing bit depth. By definition you're reducing the amount of signal present which makes your S/N worse. If by "controlling the DAC" you mean reducing the signal at the DAC's analog stage then there's nothing theoretically wrong with that (except that it's rarely if ever the case in computer sound cards or hi-fi DACs) as it is the equivalent to adjusting the volume on your (integrated) amp. Probably the most common exception to the above is people using a digital EQ to attempt to improve the sound of a mediocre computer audio setup. Most digital EQs allow for both "boost" and "cut" -- increasing or decreasing the amplitude of individual frequency bands. If you EQ the music the proper way you should only cut frequency bands in order to emphasize the other bands to suit your listening tastes. For example if you want to try to increase the amount of bass you should cut the mids and the highs and turn up the volume (on your amp) to compensate. EQing this way ensures that you're not forcing the signal into clipping by boosting too much. You still won't get great results because if you're lacking bass or highs then it's most likely because you're not running full range speakers. If the sound does change when you EQ it's most likely due to increased distortion, not because you're appreciably increasing the amount of bass present. That's not true for full range speakers but if you have full range speakers you probably wouldn't need to EQ in the first place. (Full range speaker are usually 20 - 50hz to 20khz.) If it's still not clear than think of it this way: Standalone CD players don't have volume controls. Their output will be "max volume" of what's on the CD. You can think of your computer and sound card like a (crappy) CD player. Make sure it's at max volume and adjust the level on your external amp or amplified speakers.

asynchronous13 14y ago
'As discussed, lowering the volume in software ... is accomplished by reducing bit depth.' No, reducing the amplitude of a digital signal does not change its bit depth. If the amplitude of the digital signal is reduced, then a couple of bits on the MSB side will be constant. However, reducing the bit depth would make bits on the LSB side constant. To state it differently, a lower bit depth increases the quantization levels and results in lower quality audio. Reducing the amplitude retains the same quantization level, and therefore has no reduction in audio quality (At least, not due to bit depth, it does change the SNR as you suggested)

JagMicker 14y ago

I've seen a lot of comments that seem to confuse digital and analog audio. Digital audio levels are fixed -- i.e. 0 dBFS (dB relative to digital "full scale"), thus you want the audio as close to full scale (0dBFS) as possible. This is the reasoning behind what you've probably heard described as the "loudness war". 0dBFS is the reference point beyond which no signal can exceed. Contrast that with analog audio, be it the DAC (digital-to-analog converter) on your soundcard that splits to a "line out" (likely 3.5mm a/k/a/ 1/8" TRS connector outputting audio at -10 dBV, the "consumer" level) and "headphone jack" (i.e. line level signal -> headphone amplifier). Once the audio signal becomes analog, you must then be concerned about gain staging. If you have a crappy sound card, you shouldn't reduce the analog output, because, in order to achieve the same loudness, you will have to increase the gain on your 'speakers' (a/k/a/ "pre-amplifier"; often integrated into cheap 'all in one' computer audio playback systems). Remember that the noise floor remains the same regardless of gain settings, so by reducing the "line out" level, you are only increasing the noise floor (i.e. level of noise). For all of you geeks out there... If you are looking to cut through the bullst and get a quality playback system, look into a setup that has a quality DAC and/with monitor controller, a separate preamp, and quality monitors. Benchmark has a great USB DAC, and or Lavry Engineering is what the professional mastering studios use (their DAC costs well over $20k, and not because of crap like 'voodoo diamond dust covered gold-plated platinum wires', but, in part, because it has an internal oven and temperature monitoring circuitry to keep the internals at precise temperatures). You can get a great monitor controller from Dangerous Audio or Coleman. Bryston preamp's are very popular in the professional setting, as are PMC and Bowers & Wilkins monitors.