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  • High Resolution Audio Defined!

    So now we know!

    From a press release from the Consumer Elextronics Association:

    "LOS ANGELES (June 12, 2014) – DEG: The Digital Entertainment Group, in cooperation with the Consumer Electronics Association (CEA)® and The Recording Academy®, announced today the results of their efforts to create a formal definition for High Resolution Audio, in partnership with Sony Music Entertainment, Universal Music Group and Warner Music Group.

    The definition is accompanied by a series of descriptors for the Master Quality Recordings that are used to produce the hi-res files available to digital music retailers. These can be used on a voluntary basis to provide the latest and most accurate information to consumers.

    The descriptors for the Master Quality Recording categories are as follows:

    MQ-P - From a PCM master source 48 kHz/20 bit or higher; (typically 96/24 or 192/24 content)

    MQ-A - From an analog master source

    MQ-C - From a CD master source (44.1 kHz/16 bit content)

    MQ-D - From a DSD/DSF master source (typically 2.8 or 5.6 MHz content)

    To further expand the High Resolution Audio initiative, The Recording Academy, the DEG and the CEA are sponsoring a special High Resolution Audio Listening Experience event, which will be held at Jungle City Studios in New York on Tuesday, June 24 from 6PM to 9PM during CE Week."

    So, dig out your old 4-track Portastudio(tm) masters, mix them to cassette, and call them "High Resolution MQ-A." Maybe there's more to this as they didn't specify the delivery format, only the source. Is a 64 kbps MP3 stream of a DSD recording still MQ-D? Remember when CDs were AAD, ADD, and DDD? How long did that last?
    --
    "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
    Drop by http://mikeriversaudio.wordpress.com now and then

  • #2
    CEA, after all. Just a reaction to market chatter.

    If people really knew what went on inside the sausage factory...
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    • #3
      I thought hi-res was defined as going to 11.
      N E W S O N G ! To Say 'No' Would Be a Crime (Remix) is now streamable from my YouTube channel.

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      • #4
        Wouldn't it be -2,147,483,647?
        Last edited by blue2blue; 06-15-2014, 10:57 AM.
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        • #5
          I just figured something out -- something that might actually just be sort of significant...

          If there's a push to 20 or 24 bit files, hopefully we'd see a push to include at least 48/24-capable pathways in consumer devices.

          Now, that won't help the crappy analog electronics and speakers in these things, of course, but it would presumably at least elevate currently 16 bit consumer digital electronics above the problem of [digital] volume control crap-out, where turning the level down from the digital side means the signal falls into the digital noise floor more readily.

          Now, certainly, I understand that signal exists under the digital noise floor, but it's not necessarily pretty down there -- which can get to be an issue when you plug such a device into a really loud playback system and then find yourself having to cut level from inside the digital pathway by 50 or 60 dB. 60 dB off ~140 leaves you with pretty good signal. 60 off the ~90 dB SNR of 16 bit takes you back to the 1950's.

          I have a pair of 200 w/ch powered monitors. Even with their 20 dB input pads fully engaged, I have to lower my signal by 25-30 dB or more just to keep it at a comfortable level with well mastered, unsquashed material. I elect to do it with an analog mixer (really just using it as an active volume control), which, despite added noise, allows me to put a relatively healthy signal out the DAC -- at the typical mixer level I'm still invoking as much as 20 dB of attenuation, but that's still keeping the digital signal path SNR well above that of the analog channel.


          (Of course, there's been nothing to stop consumer electronics companies from upping their digital pathways to 20 or 24 bit -- and some may well use 18 internally already -- independent of the bit depth of whatever the 'standard format' of the moment is -- and that, 'unilaterally,' would help ease the device past the 'shrinking SNR' issue implicit with digital 'volume' control.)
          Last edited by blue2blue; 06-16-2014, 12:56 AM.
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          • #6
            Originally posted by blue2blue View Post
            I just figured something out -- something that might actually just be sort of significant...

            If there's a push to 20 or 24 bit files, hopefully we'd see a push to include at least 48/24-capable pathways in consumer devices.
            I know I mentioned this before but Computer Manufacturers like HP have been including 24/96 cards in their computer for several years now. The hardware capability is there. The question really is will anyone bother taking advantage of the capability.

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            • #7
              Originally posted by blue2blue View Post
              it would presumably at least elevate currently 16 bit consumer digital electronics above the problem of [digital] volume control crap-out, where turning the level down from the digital side means the signal falls into the digital noise floor more readily.
              I didn't think anyone used the volume control any more. Isn't that why hey keep making music louder and louder? So you won't have to turn the volume up for a "quiet" song?

              Now, certainly, I understand that signal exists under the digital noise floor, but it's not necessarily pretty down there -- which can get to be an issue when you plug such a device into a really loud playback system and then find yourself having to cut level from inside the digital pathway by 50 or 60 dB.
              This is a system problem - the "really loud playback system" isn't set up correctly. Most of us understand not to connect a line level source to a mic input. When we have to do that, we should be attenuating the analog signal. In a pinch (when you don't know what you'll be connecting to until you get there) you do what you can to make it work. I gave kudos to a mixer I reviewed a while back for setting the gain of the "tape" input (RCA) jacks closer to 0 dBu than the nearly universal -10 dBu. The typical assumption is that what you'll be connecting to those inputs is a "consumer" device with a fairly low maximum output level. But today what's most likely to be connected is an iPod's headphone output, which is nominally several dB "hotter" than a 20 year old CD or cassette player.

              I have a pair of 200 w/ch powered monitors. Even with their 20 dB input pads fully engaged, I have to lower my signal by 25-30 dB or more just to keep it at a comfortable level with well mastered, unsquashed material.
              That sounds like kind of an unusual case, which calls for another 20 dB pad in line between those speakers and your source. I've encountered a similar problem in the other direction. The maximum input level of my Korg MR-1000 recorder, even in the low gain setting, is +18 dBu. Even though you can turn the record level control down to get the meters on scale, if what's hitting the input peaks above +18 dBu, the input stage, which is ahead of the record level control, will clip. I have a pair of cables into which I've built a 10 dB pad that I carry when taking the recorder out to record from a PA console where I don't have control of the source level. I also have a pot in a box that's a little clumsier than cables but gives me more flexibility.
              --
              "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
              Drop by http://mikeriversaudio.wordpress.com now and then

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              • #8
                Well, most powered monitors aren't so over-powered, for sure. Frankly, I have no idea what the boys and girls at Event were thinking when they made the 20/20bas. I firmly believe 120 W/speaker would have been sufficient for most folks. (As it is, they're bi-amped with 135 w to the woofer and 65 to the tweeter.) I guess they were thinking that many smaller studios don't have client-impressing soffit-mounted monster monitors and so they should make these things ridiculously loud. I've never taken them to the point of obvious distortion and I hope I never do. They are loud.

                (Event later came out with, as I recall, a 100w/side 'junior' version. They also had a passive version you could drive from conventional 2 channel stereo amps. Not sure if they had connections for biamping; but I think a large part of the appeal of the bi-amped active boxes is the active crossover carefully tuned to the speaker, which is presumably why they're able to deliver frequency response they spec at 38-20k Hz +/- 2 dB. On the downside, like all ported reflex speakers, their damping [lack of resonance, accurate time response] is not nearly as good as with acoustic suspension speakers like the ubiquitous NS10's. That said, if I could only keep my NS10s OR my Event 20/20bas, I'd take the Events every time, even if the Yamahas are worth double what I paid and the Events are probably worth half. )

                Of course, I could use a passive monitor controller, but the frequency imbalances that arise from passive attenuation are more a concern than the relatively tiny noise native to the (original) Mackie 1202 I use as their throttle. (Of course, I could get an active controller but, TBH, I've heard so many people going on endlessly about whether/how much their Big Knobs and Central Stations affect the sound that I'm content sticking with the devil I know, which, of course, I already have, and which I'm fairly confident of.)


                But, for sure, if one had more 'reasonably' powered powered speakers, maybe in the 80-100w range and with at least some variable attenuation and was feeding with a 24 bit system, using a control in the 24 bit path, then he should mostly be staying up in a reasonable range with normal listening levels. (And, you know, when you're turning it WAY down, you probably are willing to lose some detail, anyhow, since you're likely doing something else like conversing or such.)
                Last edited by blue2blue; 06-16-2014, 11:39 AM.
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                • #9
                  Originally posted by blue2blue View Post
                  Well, most powered monitors aren't so over-powered, for sure. Frankly, I have no idea what the boys and girls at Event were thinking when they made the 20/20bas. I firmly believe 120 W/speaker would have been sufficient for most folks. (As it is, they're bi-amped with 135 w to the woofer and 65 to the tweeter.) I guess they were thinking that many smaller studios don't have client-impressing soffit-mounted monster monitors and so they should make these things ridiculously loud. I've never taken them to the point of obvious distortion and I hope I never do. They are loud.
                  It's not the watts that make them appear loud in your studio, it's the number of volts going in that's required to get to that power level - in your case, not many. There's a lot of voltage gain. For quite a while, I was using a Hafler DH-120 to drive my (passive) speakers. That amplifier has a fixed input level and when my Soundcraft mixer (or any "standard" mixer, for that matter) was showing 0 VU, I got plenty of volume with the control room monitor level control on the mixer at about 9 o'clock. Anything beyond 12 o'clock would clip the input of the amplifier. I wanted to have more working range on the control room monitor knob, so I just added a 12 dB pad to the input of the amplifier and now I can turn the mixer up to 11 without clipping or driving myself out of the room.

                  That's "gain staging."

                  Of course, I could use a passive monitor controller, but the frequency imbalances that arise from passive attenuation are more a concern than the relatively tiny noise native to the (original) Mackie 1202 I use as their throttle.
                  What frequency imbalances? Nothing has flatter frequency response than a resistor. What you can get with a passive volume control, and really, what's in your Mackie mixer is the same thing, is a slight change in the left/right balance over the rotational range of the pot. Your off-the shelf dual pot just isn't made for really accurate tracking between the two elements. It gets worse if you're building a balanced stereo volume control because then you need four pots and if the two variable resistors on a channel don't track accurately, the common mode rejection will suffer.

                  Many years ago, I wrote an article in Recording Magazine about how to work out a do-it-yourself project, and used a monitor controller as an example. I think I may have "invented" the monitor controller with that article because at the time there was no such thing. The article was about how to define a problem and figure out what it takes to solve it. To solve the tracking issue, I proposed buying a handful of dual pots from Radio Shack (they only cost about $2 at the time) and showed a test setup using a battery and a voltmeter to pick out the one of the batch that had the most accurate tracking between elements.

                  --
                  "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
                  Drop by http://mikeriversaudio.wordpress.com now and then

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                  • #10
                    Hmm... I have a rule that I generally defer on electronics issues to those who conclusively know more than I do -- which you conclusively do -- but I have to say that I did an hour or two of research on passive volume controls a year or two ago and it was certainly my takeaway that passive attenuation networks result in impedance shifts across their attenuation range that result in tonal imbalance. But I'll get back to you on that -- don't rewrite any manuals on my account.


                    Specifically, I'll be taking a good look at this article from the nice folks at Benchmark: http://forum.benchmarkmedia.com/disc...l-technologies

                    Here's what they write in the intro to their section on passive volume control:

                    Passive Attenuator

                    A passive attenuator is simply a resistor network or potentiometer creating a voltage divider in the signal path. The output of a voltage divider is a scaled version of the input signal. A passive attenuator uses only ‘passive’ components, which are components that do not require a power source.

                    Passive attenuators have a reputation of being completely benign. However, a poorly designed passive attenuator can be detrimental to the quality of the audio. Passive attenuators can add noise and distortion, and they often change the frequency response of the system. Passive attenuators with high impedance (greater then 500 ohms) are particularly problematic.
                    [bold added]

                    That 'poorly designed' is obviously a potentially very important qualifier.


                    EDIT: I'd recommend that Benchmark whitepaper on volume control to anyone hazy on volume control issues since it explains them well at a level most reasonably experienced recordists should get right away. The section titled, Overview of Volume Control Implementations, does a much better job of exploring the digital volume control issues I touched on.

                    In addition to dynamic-range limitations, one should also be concerned about distortion induced by an inferior DSP algorithm. If the designer does not implement proper dithering, severe non-harmonic distortion will occur. Many computer playback systems lack dither. 16-bit systems have noticeable distortion when dither is omitted. 24-bit and 32-bit systems are much more forgiving when dither is omitted.
                    Of course, we all carefully apply dither in our DAWs (or more likely our DAWs are set up to do it automatically at our discretion) when performing DSP or truncating bit depth of signals.

                    But if the digital volume control in our computers' media playback systems, our phones, and various consumer playback devices do not properly apply dither, on top of the reduced SNR, we not only get greater alias distortion -- but our signal is that much more 'submerged' in that distortion and noise floor and, consequently, such distortion becomes that much more noticeable.
                    Last edited by blue2blue; 06-16-2014, 12:28 PM.
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                    • #11
                      I suppose that a passive attenuator could cause a high frequency loss in a poorly designed system - assuming that the system included the cable between the attenuator output and destination input.

                      Given that a typical line output from a solid state device has a source impedance of 50 to 100 ohms, you don't want to load it with anything less than ten times that impedance. So if your attenuator consists of a 1kΩ shunt resistance and you want 10 dB of attenuation, that means your attenuator will consist of two resistors, about 700 ohms and 300 ohms, and you'll take the output to the next device across that 300 ohm resistor. So the source impedance is now 300Ω instead of 50 or 100Ω.

                      That's not too bad, but let's say you use a 100kΩ pot so you can have a volume control. Now, for 10 dB of attenuation, you have a source impedance of 30kΩ. That's getting into the range where the capacitance of the cable can start making it act like a high-cut filter if you have a long enough cable. And since the source impedance varies as you move the slider on the pot, the roll-off frequency of the filter you've made by adding a piece of cable can change. Surely you've heard the arguments about using high-juju guitar cables when you want to stand more than a few feet from your amplkifier (or, from companies who make such cables, even if you want to sit on your amplifier while you play).

                      So, like most anything else electrical, the effect is a matter of degrees, and it can be measured. Benchmark has products to sell that cost more than a Radio Shack pot or a couple of fixed resistors. You need to decide if the improvement is really significant enough to be worth the cost. For some it is, for the rest of us, it usually isn't.

                      .
                      --
                      "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
                      Drop by http://mikeriversaudio.wordpress.com now and then

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                      • #12
                        Well, I only used the Benchmark page because it was pretty well laid out and had a good overview -- but I've read about this issue a fair amount in materials going back years. This info and the conclusions are hardly unique to Benchmark. But I'll see what more I can come up with.
                        Last edited by blue2blue; 06-16-2014, 01:40 PM.
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                        • #13
                          Why don't you just try it and see what it does to your system? We've used passive attenuators much longer than we've agonized over high resolution audio. Probably all of your favorite recordings have them somewhere in the production chain.
                          --
                          "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
                          Drop by http://mikeriversaudio.wordpress.com now and then

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                          • #14
                            Well, like I said, I'm already using an active control, so I have no problem to be solved. With regard to any questions I have about whether or not frequency response linearity can be affected by a passive volume control network inserted at line level, I really didn't have any question before your first post on the issue, based on my reading as well as at least a little hands-on experience. (That said, obviously you have to have some gain-staging flexibility to properly A/B at the same listening level, and it can be risky generalizing from limited experience with possibly far-from-exemplary devices.) But, like I said, by rule I defer to people I know have greater knowledge and expertise in the field of discussion.
                            Last edited by blue2blue; 06-16-2014, 03:54 PM.
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                            • #15
                              Originally posted by blue2blue View Post
                              Now, that won't help the crappy analog electronics and speakers in these things, of course, but it would presumably at least elevate currently 16 bit consumer digital electronics above the problem of [digital] volume control crap-out, where turning the level down from the digital side means the signal falls into the digital noise floor more readily.
                              Thanks for posting the only valid criticism I've seen of 16-bit systems. (I'm only discussing systems where the audio is audio-in, audio-out, not heavy FX and mixing applications.) For my mind, 16/44 sounds as good as anything, and nobody has made a strong case showing that it's not true. But this is definitely an issue in any system where the master volume control is digital rather than analog! Even my mud-standard ears hear this quite clearly, and it reminds me of when I was doing 12-bit sampling and how the tails sounded. Or of my trusty old Ensoniq MR76 keyboard, which I really like but which has a digital master volume, sadly.

                              Of course, most modern low-cost gear is this way. (I did have some laptops where I could tell that the volume control was analog, oddly, but I got out of the habit of using them at anything below nearly full tilt long ago.) So, it's a relevant point.

                              I'm confident that MIke is correct about passive level controllers. After all (unless things have changed radically), an active volume control" is just a passive one followed by a gain stage, in virtually all mixers and consumer audio gear. No doubt there are significant exceptions, such as in a mic preamp, and possibly in super hi-end preamps like a Mark Levinson.

                              Originally posted by Craig Anderton
                              Is a 64 kbps MP3 stream of a DSD recording still MQ-D?
                              I'm confident that the "source" would be the weakest link prior to hitting the hi-res digital form, so sorry, no soap.
                              Last edited by JeffLearman; 06-19-2014, 11:47 AM.
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