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Testing audiophile claims and myths

Discussion in 'Sound Science' started by prog rock man, May 3, 2010.
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  1. gregorio
    I've never come across that unit before. Obviously it's a consumer unit due to the connection type, plus I can't think of any professional situation where it would be of any use. Standard procedure when testing pro AD/DA units is to loop-back the stereo signal 10 times, in order to make the difference audible! With a relatively cheap consumer unit, especially as it's level appears to be considerably lower, fewer loop-backs might do the trick. I'm not sure what's causing the level drop by the way, although most likely it's one of two possible explanations: 1. It's reducing the level to give itself headroom for adding DSP or 2. You're feeding it a single-ended input, which would be roughly 6dB lower than the balanced input it's expecting.

    There's not really a general set of audio fragments. We (audio engineers) each tend to have our own individual sets, which comprise a number of tracks (or bits of tracks) we ourselves have worked on and therefore know intimately what they should sound like and what problems we've had to address. The situation is more tricky for consumers, they don't know what/where/if there are problem areas and they don't know what any particular track should sound like. And again, usually we've got either a very good idea of what we're trying to detect (say from measurements) or a fairly good idea of what we're trying to detect. If for example we're comparing speakers, we might use a different set of test fragments/tracks than if we're comparing AD/DAs, because the artefacts are going to be significantly different.

    For the consumer, it wouldn't but professionally it could. Particularly with non-acoustic music genres (and TV/Film sound), raw recordings are very likely to be heavily processed and therefore some artefact at -100dB could end-up becoming audible. A consumer should never encounter this situation though, unless they have some pretty serious flaw with their setup (very poor gain staging for example) but then of course they should address that flaw, rather than looking to improve something that should be well below audibility.

    castleofargh likes this.
  2. KeithEmo
    There are a whole bunch of little "DSP boxes" with a variety of purposes. The popular "Mini DSP" boxes, which are really just general purpose audio DSP boxes you can load a variety of software on, can be loaded with a variety of filters and software - including crossover filters, room correction EQ, and even fancier room correction apps like Dirac Live. They come in different versions, with different numbers of channels, and balanced, unbalanced, or digital inputs and outputs, and offer a variety of "software modules", and the option of designing your own filters and downloading them to the box.

    The input and output levels on consumer equipment tend to be somewhat arbitrary... You aren't looking at "a level drop" per se. The maximum level of a digital signal is specified. What you're seeing is simply the maximum input level of the ADC side, and the maximum output level of the DAC side, both of which are independent of each other, and are determined by the analog circuitry in the box. (In general, the designers will have picked an input level such that "it's unlikely to be overloaded by a typical audio source" and an output level that's "plenty to drive whatever you may want to connect it to".

    I have always found that it's best if YOU pick out a set of music to audition equipment with that is both demanding and familiar to you. I have never had great luck trying to determine how equipment sounds, or whether I can hear a difference between two different things, unless I am first extremely familiar with the sample music I'm using. That way I know what it should sound like, what specific things to listen for, and exactly what to expect.

    On a song I am familiar with, I recognize when a particular guitar pluck isn't quite right, or when I can't hear the scrape of a certain chair between two words in a certain song, or when the hall ambience isn't quite right, or the sound of an intake of breath at the beginning of a certain line in a song ... and I have no chance of noticing those sorts of details in a recording I'm not familiar with.

    Most of us also have a tendency to become acclimatized to a new piece of music over the first several times we hear it. We may notice more details as we listen each time... we may hear new details that make us like the track more... or we may notice annoying details that cause us to quickly become tired of it. To me, both of those situations constitute a bias, which cause my opinion of that track to "drift" over time. And, as long as that's still happening, even when I play it on exactly the same equipment, "each time I play the track it's a little different", which makes it impossible to compare different equipment... until I am what I would consider "totally familiar" with the track I'm using to compare it. (It's like the difference between trying to compare a celebrity you've never met in person with a professional impersonating them... and comparing someone you've known for a long time to someone trying to impersonate them at dinner. In the second instance, not only are bogus details likely to be more obvious, but you're also more likely to recognize that "it isn't my old friend"... even if you can't point out the specific details that you noticed.)

    It's also worth noting that the music I may choose to listen to when evaluating equipment is not at all the same music I would use to show off that equipment. When auditioning gear, you should always bring your own sample music, rather than rely on what the salesman, or your buddy who just purchased it, prefers to use to demo it. This is true for several reasons. First, you want something you're familiar with. Second, the factors that you consider important may be different than those your buddy considers important. And, third, obviously the salesman is going to pick music that "plays to the strengths, and avoids the weaknesses, of the gear he or she is trying to sell you". (For example, when the Advent loudspeaker was first offered for sale, EVERY audio store I know demoed them with a tune called Dance With Me by a group named Orleans.... because those particular speakers sounded just awesome with that tune.)

    Since the original request was for suggestions... I'll throw out a few...

    If you like bass, and organ music, try: The Six Wives of Henry VIII - by Rick Wakeman
    And, if you like acoustic guitars, try: The Eagles - Hotel California

    Both of those have been re-mastered several times....
    So use a specific version with you that you are familiar with.

    kukkurovaca likes this.
  3. sander99
    That's what I am doing indeed.
  4. gregorio
    Yes, I'm aware of some of the "Mini DSP" boxes, just not the particular one mentioned.

    There are international standards for unbalanced (consumer) line level and balanced (professional) line level, although I agree that many/most consumer units don't adhere to the standards. The actual difference between the two should be 11.8 dBu but typically with consumer gear the balanced connection is roughly 6dB hotter.

    1. That was my point though, you DON'T know what it should sound like! You know what you are familiar with (and what you prefer) but your reference is arbitrary, your personal sound system/s.

    2. Again, how do you recognise that a particular guitar pluck, hall ambience or intake of breath "isn't quite right"? I don't doubt that you can hear a difference but which one is "right" or more/less "right"? "Right" is just a reference to your own system/s and personal preferences. I've never heard a consumer/audiophile system which sounds similar to my studio/system and whenever I've had an audiophile in my studio, they've ALWAYS commented how surprisingly different it sounds and typically (though not always) they've preferred their own system. If I've created the music, then "Right" is how it sounds in my studio but audiophiles pretty much never get to experience this actual "right", their "right" is just a subjective personal preference (on systems which are ALL significantly different from "right").

    That's going to make a direct AD/DA comparison difficult, especially if you're doing loop-backs, as with each loop-back you're cumulatively increasing the noise floor (and any artefacts it contains) by an additional 7dB. So effectively you are comparing gain-staging rather that the AD/DA process! I'm not sure exactly how you'd alleviate that problem, personally the first thing I'd try is: Do say two loop back recordings WITHOUT your 7dB gain on each loop, reduce the original by 14dB and then boost them both to the same volume level with your amp to compare them. Remember though, you're only listening for differences NOT making any quality judgements, because the quality of both will be degraded (a 14dB higher noise floor just for starters). I'd measure the actual levels though, rather than just guessing at 7dB and I'd also perform a null test.

    Last edited: Oct 29, 2019
  5. sander99
    I guess this means I would always throw away 7 or 6 dB signal to noise ratio when using these boxes unbalanced? Not that that's necessarily a big problem for me.
    I already did some listening with 8 loop-backs and indeed noticed the added noise.
    Well, it is what it is, I can not test them under optimal conditions but I also can not use them under optimal conditions so in the end for me the question is: will they do good enough under the current conditions?
  6. KeithEmo
    I agree entirely... to a point.

    You have no absolute reference.... but at least you have a relative reference.... and that works for you in two ways.

    If you have some favorite music that you always use when auditioning equipment then, even though you may not know what that music sounded like in the recording studio, at least you know what it sounded like on a lot of other equipment. As neuroscientists love to point out, we can't really know if I see the same thing as you do when I look at something "red". However each of us has built up an internal reference of "what red looks like" by looking at lots and lots of things that we have been told are red. Likewise, if you listen to a certain set of familiar songs on a lot of equipment, and make sure to include a lot of equipment that is generally considered to be "good", you can build up a decent "consensus average" of what it should sound like. In fact, if you pay careful attention, you can even build up a detailed idea of "what it sounds like on gear that most people say is bright" and "what it sounds like on gear that most people say is harsh". It's not perfect.... but it's a lot better than no idea at all. (And, if you're lucky, you may actually have an opportunity to listen to it in a studio someday.)

    As some other folks like to point out... this thread is largely targeted towards people making purchasing decisions. In that context, from a purely relativistic perspective, if you know what familiar music sounds like on your system, you also probably have a pretty good idea of what you do and don't like about how it sounds..... This gives you a good basis for deciding whether it sounds "better or worse" (to you - based on your subjective opinion) on new gear you may be auditioning. Also, to put it bluntly, if you're purchasing new equipment, one of your goals is probably to improve how your system sounds with the music you listen to. (There's not much point in auditioning a system, and choosing one that sounds really great with jazz music, if all you listen to is heavy metal... after all, there's no guarantee that every system that sounds good with jazz will sound good with heavy metal, and most consumer systems are not going to be "extremely accurate with all types of music".)

    The alternative here is to use music which you are NOT familiar with, and so have no idea whatsoever what it should sound like, in a studio OR on your personal system.
    Of course, the other alternative is to choose your equipment based solely on measurements, but that often seems not to work out well...
    (And that's especially true for non-pros who don't have a solid basis for interpreting what the measurements mean.)

    Dani157 likes this.
  7. gregorio
    1. Not necessarily. If you've got the box in your playback chain but not actually applying any DSP then yes, you're effectively throwing ~6dB away but if you're applying DSP, say a +6dB EQ boost, reverb or some other process which adds to the signal, then effectively you're not really throwing anything away.

    2. The problem you have is that you don't know if/how much of that noise is due to the AD/DA process and how much is due to gain staging. Given a theoretically perfect AD/DA process and a theoretically perfect test recording (with an infinitely low noise floor), each loop should result in very approximately 6dB more noise (due to dither and thermal noise) but presumably you're just using a music recording, which has a high noise floor relative to dither + thermal noise, which you're boosting by 7dB per loop. This boosted recording noise floor likely accounts for most (or likely virtually all) of the added noise you're hearing.

    1. A lot of equipment that is "generally considered to be good" is setup poorly, for example, speakers that are "considered good" placed poorly or in a poorly shaped room with little/no or inappropriate acoustic treatment. Additionally, in the audiophile world "considered good" and "accurate" are often two quite different things (which are conflated). So in general I disagree, the average consumer (or audiophile) cannot build-up a decent consensus average of what it should sound like. Again, I've never heard an audiophile system which sounds like my studio, sometimes/often deliberately so. Pretty much without exception, every audiophile I've had in my various studios for nearly 30 years described the experience as highly analytical (typically shocking so) and/or some variation of "unmusical".

    2. Firstly, I agree ... but obviously, as you state, that's "better or worse" relative to what an individual is familiar with and what their personal preferences are, not intrinsically or objectively "better or worse". Commonly, what is "better" subjectively to a particular audiophile (or even group of audiophiles) is actually "worse" but nevertheless they'll describe it as (or imply that it's) actually "better". This is the root cause of many audiophile myths! Secondly, what one is familiar with (and one's preferences) can change, sometimes dramatically and unpredictably. It's quite amazing how human hearing/perception can adapt to a very different system (with familiarity) and preferences take an almost 180deg turn, especially if they believe the different system is at least in some way better/more accurate. I've seen this several times with audiophiles I've worked with (inexperienced directors or acoustic musicians unused to recording studios), who initially hated the sound in my studio compared to their audiophile system.

    Most consumers/audiophiles of course don't have much choice and I agree that using music they're familiar with is better than music they're not. I'm just pointing out the trap many fall into, namely confusing/conflating: Knowing what it should sound like with knowing one's preference for what it should sound like, there's typically a significant difference between the two!


  8. sander99
    I hope I don't bore anyone to death with this long post... only read on your own leisure, if at all...
    (A detail I left out first is that I didn't have all the boxes outputs at +7 dB, that was a first (and too high) rough estimation made listening with one pass through, at the last outputs of the 8 times loop I had a smaller value to correct. I now have +6.2 dB in each output, which is a tiny bit too low. I still have to do a precise measurement. But at least going from 8 to 1 loop the error is divided by 8 also.)

    DSP 4x4 Mini input specs: Level +12 dBu, Impedance 1 MΩ (stereo), 500 kΩ (mono)

    According to this calculator:

    +12 dBu means 3.08 Vrms / 4.36 Vpeak

    Would that be (1) the voltage difference between hot and cold, or (2) between hot and 0?
    Then inputting an unbalanced signal with 1.54 Vrms / 2.18 Vpeak would be
    In case (1): +6 dBu
    In case (2): 0 dBu

    But to minimise the degrading effect of the ADC wouldn't it be better to input max level, and lower the input gain - assuming this works in the digital domain after the ADC - to create headroom for the processing?
    (Internal processing is 32 bits by the way.)

    Actually I have listened to the boxes in 2 different situations:
    1. the boxes after the volume control (between the pre-outs of my receiver and the main-in power amp inputs of an integrated amp with removed pre-main bridges). Which is of course the worst possible situation.
    2. the boxes before the volume control (between source and a normal input of the receiver).

    As a "source" I used the zone 2 output (fixed volume, so full line level) of my Yamaha RX-V771 receiver but it is: 200 mV (1.2 kOhm); which according to the calculator is: -11.76 dBu, so almost 24 dBu too low!
    Still very far from optimal.
    [Edit: and this was in case (1), but now I know case (2) is "the case", so the situation is even worse: I am inputting almost 30 dBu too low! Which actually increases my respect for these boxes because except the added noise I can not hear anything wrong with 8 loop-backs.]

    My intended use for the boxes is between the 16 analog outputs of my Smyth Realiser A16 (on order) and 16 channels of amplification in the form of 2 obsolete Yamaha RX-V750 7.1 receivers with analog 7.1 inputs and an old Yamaha AX-592 integrated stereo amp. For use of the A16 as a 16 channel decoder with real speakers, and for PRIR measurements both with PEQ, level, and delay in the chain which is otherwise problematic to achieve with only the analog outputs available on the A16 (most av receivers can not apply any dsp to analog 7.1 inputs).
    Ideally I would have to do the volume control after the boxes (set A16 volume to max, or max -2.24 dB, see below), maybe I should find a third RX-V750, because their volumes follow the remote control beautifully synchronous (with digitally displayed discete steps, whereas the AX-592 has a motorized potentiometer that of course doesn't follow the others correctly and even by hand would be difficult to set right).

    The A16 16 channel analog output specs are a bit more promising:
    Impedance 10 Ohms
    Peak Output 2 Vrms
    Which would mean +8.24 dBu.

    So in case (1) at least I can reach the +6 dBu on the DSP 4x4 Mini inputs.
    But in case (2) I can reach only +2.24 dBu.

    But I am wondering if in case (1) it would be possible to input +8.24 dBu in the box unbalanced? If the input was implemented using a transformer (very unlikely I guess), or a differential amp that doesn't care about the absolute voltages (if that is possible at all?)...

    By the way [Edit: back to the 8 loops test situation]: if I only had the cables I could do the actual loop-backs themselves balanced, so that only the first inputs and the last outputs of the loops are unbalanced. And I could raise the input gain of the first inputs (and lower the output gain of the last outputs) such that the signal reaches full level and probably stays that all the way through the 7 "middle" DA-AD (from the inside-the-box perspective) conversions. Then I could check the influence of 8 conversions with only one time the (extra "gainstaging") added noise. I have the first inputs and the last outputs in one and the same box, so in that box I could switch (using the "Matrix", just unfortunately not a fast instantanious switch) between including or excluding the additional 7 DA-AD conversions (all full level, balanced connections).
    Last edited: Nov 1, 2019
  9. sander99
    Ha ha, maybe not so smart to put "only read on your own leisure, if at all..." above my last post because now nobody answered this question:

    Generally, if an input level for a balanced input is given, for example:
    +12 dBu, which means 3.08 Vrms / 4.36 Vpeak
    Would that be (1) the voltage difference between hot and cold, or (2) between hot and 0? (Which would make a 6 dB difference).
  10. sonitus mirus
    European studio level is +6 dBu, which is 4.38 Vpp (2.19 Vp) and 1.55 Vrms, so +12 dBu is peak to peak (8.7 Vpp).
  11. gregorio
    I'm not sure I understand and without the other piece of the puzzle, it doesn't mean much. The basic principle of a balanced connection is that the voltage between hot and cold is identical but inverted (out of phase). At the balanced input the cold signal is inverted (in phase), giving us two identical signals which are summed, effectively resulting in signal level x 2 (+6dB). Feeding this balanced input with an unbalanced signal means there is only one signal, not two and therefore the level is 6dB less.

    Studio/Pro audio line level is +4dBu, while consumer line level is (supposed to be) -10dBV, which equates to -7.8dBu. However, what's important here is what that analogue signal level is referenced/calibrated to on the digital scale (dBFS), the missing piece of the puzzle. This is fixed by (and adhered to) international standards in the case of theatrical film and HDTV at: +4dBU = -20dBFS (peak level is therefore 0dBFS = +24dBu). There is no international standard for music though, by convention it's commonly +4dBU = -16dBFS but can be anywhere between -12dBFS to -20dBFS. Is that quoted +12dBu, the peak input (0dBFS)? If so, that's 12dBu lower than pro line level but if it's say -20dBFS then it's 8dBu higher than pro audio line level. In addition, do you know where the 6-7dB attenuation is actually occurring in your loop back? Maybe it's on the output?

    More analogue gain usually incurs more internal self-noise (depending on it's nominal gain design), so having a lower analogue gain level would result in lower noise than a having a higher analogue gain level and reducing the digital gain. In fact, that's pretty much the whole point of 24bit A/D conversion, to allow for a lower analogue signal (mic pre-amp) level and therefore significantly more headroom, without having to worry too much about clipping or the noise floor of the digital domain. In this case though, the nominal design is for a higher level (balanced) analogue signal. So, assuming basic competency in the analogue input design, then it wouldn't make any audible difference, you'd probably be hard pushed to even measure a difference, maybe just a bit of extra thermal noise way down around -130dBFS or so.

    Last edited: Nov 2, 2019
  12. sander99
    That I understand. And that's why <the voltage difference between hot and cold> is 2x <the voltage difference between hot and 0>.
    Example: at one point in time hot = +2 V. cold = -2 V.
    <The difference between hot and cold> = (+2)-(-2)) = 2 + 2 = 4 V,
    <the difference between hot and 0> = 2 - 0 = 2.
    Maybe it looks a bit silly to write it down like that but it's just to show what I meant. Also I see now how my original question could be misinterpreted because the max amplitude of a signal is always a positive value of course, so the difference between the max amplitude of hot and the max amplitude of cold would be zero. By the way I assume Vpeak means max amplitude? (Ignoring "the other piece of the puzzle" for a moment.)

    So to rewrite my original question a little bit:
    +12 dBu means 3.08 Vrms / 4.36 Vpeak
    Does this mean hot and cold both have a max amplitude of 2.18 V and hence the max amplitude of the hot - cold difference signal is 4.36 V?
    Or does this mean hot and cold both have a max amplitude of 4.36 V and hence the max amplitude of the hot - cold difference signal is 8.72 V?

    I understood from @sonitus mirus answer that the latter is the case, hot and cold both have a max amplitude of 4.36 V and hence the max amplitude of the hot - cold difference signal is 8.72 V.
  13. Mark74
    Yes, Vpp approx 4.3v betw Hot and Gnd as per Sonitus.

    Conventionally, rms and p-p values characterize the amplitude variation of a single signal rather than a comparison between two signals, however closely they may be related.
    Last edited: Nov 3, 2019
    sonitus mirus likes this.
  14. castleofargh Contributor
    I have to say that I really don't understand where you're trying to go with this. surely you've found one of the gazillion online converters like http://www.sengpielaudio.com/calculator-db-volt.htm
    are you asking if a unit is another one? if a single ended circuit is in fact secretly balanced? I confused about your confusion ^_^.
  15. sander99
    I am not going anywhere with this because my question has been answered, I just wanted to explain better to gregorio what I meant.
    You didn't get what it was all about? The DSP 4x4 Mini's I was talking about have balanced inputs and outputs (that I use unbalanced but that doesn't change the question or the answer). My question was about balanced inputs. The voltage specification could have been interpreted in 2 different ways as I explained. I briefly tried to google it before but apparently it is one of those things that everyone finds so obvious that they never mention it.
    castleofargh likes this.
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