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]]>The post What Is The Best Brand Of Microphone For Heavy Metal Performers? appeared first on Metal-Fi.com.
]]>The post How Important Is The Microphone To Audio In Heavy Metal? appeared first on Metal-Fi.com.
]]>In summary, the choice and quality of the microphone are pivotal in capturing the essence and power of heavy metal music. Investing in good microphones tailored to the specific needs of the genre can significantly enhance the recording and live performance experience.
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]]>The post How Has Dynamic Range Changed In The Last Decade? appeared first on Metal-Fi.com.
]]>Overall, the dynamic range in audio systems has significantly improved over the last decade due to advancements in digital processing, high-resolution audio formats, better recording techniques, enhanced playback devices, more careful use of dynamic range compression, and high-quality streaming services. These improvements allow listeners to experience richer, more detailed sound with a wider range of volumes.
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]]>The post Sample This Again appeared first on Metal-Fi.com.
]]>Downsampling is simply the process of reducing the sample rate of the original signal. Let’s say you originally recorded Bolt Thrower’s new album (a miracle in itself) at 24-bit/96kHz in your home studio and wanted to share its glory with the rest of the world which has standardized on the CD, what do you do? You downsample, which means you take the original 24-bit/96kHz sampled waveform and run it through a process that will re-sample it at 16-bit/44.1kHz.
So the first question that springs to mind is doesn’t downsampling reduce the overall fidelity since we are losing bits? No, provided we maintain a reasonable target Nyquist rate. In other words, by downsampling from 24-bit/96kHz to 16-bit/44.1kHz we can still reconstruct frequencies up to 22.05kHz or slightly above the range of human hearing.
But what about the newly introduced quantization noise introduced from taking all those 24-bit samples and trying to represent them now as 16-bit values, won’t that effect the music’s overall fidelity? Not really. Through a technique called dithering, we can push the noise introduced by the downsampling process to levels not audible by the human ear. Though it sounds a bit unintuitive, dithering actually adds some random noise to the original signal in order to cause the quantization errors introduced during the downsample process to be non-deterministic, causing the overall average of error within each sample to be closer to the original waveform. Still with me? Suffice it to say, you will be hard pressed to tell the difference between the original 24-bit/96kHz recording and the dithered down 16-bit/44.1kHz release.
Many high end studios record in 24-bit since within a studio context, there are some real advantages to be had. First, 24-bit sample sizes can represent a higher level of dynamic range, theoretically 144dB instead of 16-bit’s 96dB. Secondly, and even more importantly, recording at a higher bit depth yields a lower noise floor, or the intrinsic noise generated from the recording system itself. Both of these properties gives the engineer more “breathing” room during the recording process since low level noise such as note decay or very quiet passages may sound odd or distorted with only 16-bits to work with. Couple that with a complex production chain that includes dynamic range compression and brickwall limiting, the more headroom you have the better! Hear for yourself.
However, by the time the music hits your ears all the above is moot. The fact is how the engineer mixed and mastered the music is of way more importance than what sample rate he used to record it. That’s part of the reason why my partner in crime is constantly chastising hi-res distributors like HDTracks for not clearly stating the provenance of their masters. For example, I much rather have a 16-bit/44.1kHz FDR master sourced from the original analog tapes than a 24-bit/96kHz compressed nightmare sourced from the remastered brickwalled CD.
As you probably already surmised, upsampling and oversampling are the exact opposite of downsampling. Simply put, they are both used to describe the process of increasing the sample rate of the original signal. The only real difference is oversampling insinuates that the sample rate is above what was needed to meet the original signal’s target Nyquist rate. But the fact is, they are mathematically equivalent.
Please note, upsampling does not add any new information to the original signal. I will repeat (it is a very important concept), upsampling does not add any information to the original signal. So if you rip a CD, which by definition is 16-bit/44.1kHz, into 24-bit/96kHz or even 24-bit/192kHz FLAC, all you’ve done is waste hard disk space. You can’t put back what’s not there.
Related Pages:
Pale Communion High Res Smackdown
So why offer these higher sampling rates at all if dithering down to 16-bit/44.1kHz is “good” enough? There are those that feel that given today’s storage and playback capabilities, there is really no need to add more quantization noise as part of the downsampling process. Moreover, during playback, it has been shown that higher sampling rates do have some intrinsic benefits during digital audio conversion such as being more resilient to jitter. Whether or not these artifacts are really audible is another story entirely, but suffice it to say that with the inevitable death of the CD, the whole process of dithering down these days seems superfluous anyway.
Finally, when you needle drop at a higher sampling rate than the venerable CD, you are not upsampling since you are converting analog to digital, not digital to digital. All you are doing is deciding what the initial sampling rate is, and therefore what target Nyquist frequency you want to reproduce in the digital domain. Some folks like to needle drop at 24-bit/96kHz while others seem to think 24-bit/192kHz is a good idea (it’s not). But the reason why they choose a higher sampling rate is because a cartridge can produce frequencies way past the 22.05kHz boundary (or half of 16-bit/44.1kHz). Can you hear above these frequencies? Not on your life, but there is an argument to be made that if you are preserving vinyl as a digital archive you want to be able to reconstruct the original analog waveform as closely as possible.
In my next article in this series, I will throw samples completely out the window (literally), as I will be covering every audiophile’s worst nightmare, the MP3. Or is it? Stay tuned.
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]]>The post Pale Communion High Res Smackdown appeared first on Metal-Fi.com.
]]>But regardless of your feelings toward the record itself, Pale Communion‘s production is nothing short of stellar, which should come to you as no surprise given the fact that Steven Wilson was at the mixing helm. His mix not only ensures that all of the band’s proggy inklings were accurately accounted for, but also adds to the overall tenure and tone of the album as well, giving Pale Communion a sense of aural vibrancy while still maintaining its dark and brooding atmosphere underneath. Couple that with Paschal Byrne’s tastefully dynamic master, and Opeth’s latest may well be their best sounding record to date.
Perhaps knowing full well of this simple fact, the band has offered up its latest magnum opus in a variety of formats. There is of course the pedestrian Redbook CD as well as the 180g double vinyl, but there is also a 24-bit/96KHz high-res HDTracks digital download and even a 5.1 surround sound version released on Blu-ray too (an Opeth first). Today’s article is going to compare the Redbook edition to its high-res counterpart, and see what, if any, is there to be gained from spending a few extra bucks for well, a few extra bits.
First, let’s do the numbers. I took one of my favorite tracks on the record, “Moon Above, Sun Below,” as my reference sample, and threw it through Spek just to verify (as a rough cut) any spectral differences between the 24-bit/96kHz HDT version and its 16-bit/44.1kHz Redbook counterpart.
As you can see, the HDT version appears to be a true high-res release, with frequency information way past Redbook’s Nyquist of 22.05kHz, and even showing some spectral cruft near the top end around 45kHz or so. Though spectral analysis can be misleading, as far as I’m concerned, the 24-bit/96kHz is exactly that, with no dubious upsampling or odd ball lowpass filtering applied.
Now from a purely loudness perspective, both of these tracks are nearly identical. For our sample track, both clock in at DR11 and both peak at -0.10db with an RMS of about -13.45dB (for those sticklers out there, the HDT version measures -13.44dB on the TT). In English, level matching via ReplayGain or some other loudness equalization technology is not really necessarily since they are both already at virtually identical volume levels.
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As far as my playback chains go, I mostly switched back and forth under Audirvana Plus v2.0.2 using my Jerry Harvey Audio Siren Series Roxanne CIEMs through a LH Labs Geek Out IEM edition (review will be up soon). I also tried listening with my HM-901 with Balanced Card as a source just to see if an additional ESS SABRE 9018Ms would yield any audible differences over the Geek Out’s “paltry” single chip version. Most of my tests were in the form of simple ABX style testing, and entirely pseudo-scientific – pretty much the defacto standard methodology of every popular audiophile review site on the Internet. Caveat emptor.
To make a long story short, I couldn’t hear a difference. Not a one. This record is so well mixed and mastered, the higher bit and sampling rate are all for naught on my setups. I’m sure there is an audiophile out there who will claim my reference playback chain wasn’t “reference enough,” but regardless, both versions sound equally fantastic to me and you can’t really go wrong with either of them.
I also found that the more I listened to this record, the more I appreciated Pale Communion’s production. I first thought this album was just a decent sounding release, and now I’m thoroughly convinced it’s a reference quality recording, and one that should have a prominent place in every MFi’ers collection. Obviously, if you weren’t a fan of their last one, Heritage, neither the CD nor the HDT version is going to sway you since Pale Communion is clearly cut from the same cloth. But if you do have an open mind, and dig 70s prog rock, this is one of the best sounding recordings on the planet right now and an absolute must buy.
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]]>The post Interview: Colin Marston of Menegroth appeared first on Metal-Fi.com.
]]>And a big part of last year’s success is due to New York’s own all around dynamic metal superhero, Colin Marston. You see by day, he is the bassist and/or guitarist for Krallice, Dysrhythmia, and Gorguts, but by night, he is mastering engineer of his own studio, Menegroth: The Thousand Caves, overseeing that all the metal he just recorded for those aforementioned acts sound brutally delicious. And if he isn’t currently busy with his own projects, then he is working on mixing and mastering some of the best of breed products from the likes of Woe, Vaura, and East of the Wall to name just a few.
Recently, I got to speak with Colin about Menegroth, the Loudness War, and about mastering metal in general. Check it.
MFi: For MFi’ers who only know you as a musician, can you give us an overview on how you got started in producing extreme metal and the history behind Menegroth: The Thousand Caves Studios?
Colin: I started recording when I was a teenager, running my own studio in 2004, and moved into my current space in 2006. One of the first albums I recorded for hire was the unreleased Thoughtstreams full length. That’s about as extreme as metal gets. I’m more interested in recording and mixing than mastering, but I started doing mastering out of a combination of necessity and being totally unsatisfied with the process most modern mastering follows.
MFi: Now 2013 was a big year for both you as an artist and engineer, with you either producing and/or playing on releases from the likes of Gorguts, Woe, Vaura, and East the Wall, as well as releasing your second solo album under the Indricothere moniker. So if you had to pick your favorite project of 2013, what would it be and why?
Colin: One of my favorite things I worked on last year was the Geryon album (Nick and Lev from Krallice’s duo). The music is really unique and awesome. I’ve found myself ripping them off a lot in my new Dysrhythmia songs. I’m also really proud of the recording. It’s nice to allow the bass to take up all the sonic space usually supplied by the guitars. We used a fun amp setup for that album: 2 tracks of bass split to three amps each: a Traynor TS50b into and Eden 4×10 (light gain), a 5150 into an Orange 4×12 (light gain), and a metalzone set to heavy distortion into a 70s fender twin. The combination sounds pretty massive. I really like the drum sound too, even though there’s nothing too out of the ordinary going on. I also really like the Mick Barr album with a piece written for Kronos Quartet. Who knows what that one is called… I think he changed the title a few times after the release.
Another one I was really into from last year was Tom Blancarte’s solo electric upright bass album. Side one is all bow and side two is percussive and neither sounds anything like bass. I also got to work on recordings for Cleric, Oneida, Man Forever with So Percussion, Oldest, Kevin Hufnagel, Castevet, Mutant Supremacy, Dan Peck, Radiation Blackbody (another recording I really like!), Brandon Seabrook, Imperial Triumphant, Panopticon, Artificial Brain, Mastery, and last but very much not least, Zevious (their new album is totally awesome and I really like the way the recording sounds), so it’s hard to pick a favorite. I really enjoyed the freedom and pace of working on my 2 Indricothere albums too. One I approached like the first, and the other is all keyboard improvisations – my first improv album.
Related Pages:
Pale Communion High Res Smackdown
MFi: As you know, the overwhelming majority of metal mastered today is typically brickwall limited, but all of your work that I’ve reviewed is extremely dynamic. It would seem that retaining a modicum of dynamics has become a priority in your work. Why are dynamics important in metal?
Colin: It’s not necessarily that retaining dynamics is my priority – sometimes I’m trying to greatly reduce dynamics. It’s all dependent on context. Rather than that, I’d say I’m not concerned with mastering for a destination volume. How “loud” an album ends up being compared to other records never enters into my process. I view mastering as reactionary: correcting problems, improving balance and general timbre. How much and in what way I compress or EQ a mix is always a response to the source material, allowing it to be the best it can be without masking its inherent character.
To put it another way, I think a lot of people view mastering as away to achieve some nebulous professional “standard” of volume and brightness. And that standard tends to be all about screaming at you and getting your attention right away by being the LOUDEST, MOST CUTTING, PUNCHY, and all of those other dumb words, it just make me think of commercials for monster truck rallies and laundry detergent. The main point is that this mentality of mastering has nothing to do with making the mixes sound better. It’s like taking a lovingly prepared plate of pasta, and instead of garnishing it with fresh pepper and parsley, you just dump a ton of spicy bright orange buffalo sauce over everything to make it catch the eye and hit the taste buds right away, but in that process you annihilated the original flavor of the dish.
Dynamics aren’t just important to metal, they’re important to all recordings. Dynamic level and frequency are the two components of sound, so the more you process either or both (with compression and EQ) the more fidelity – the more character – you loose. Dynamic range might be especially important for metal because it’s supposed to sound loud. Things only appear loud relative to quiet things. So if the difference in level between “no snare hit” and “snare hit” are a 10th of a db, the snare doesn’t sound like it’s being hit hard. If you leave some difference between off and on, you hear an attack – the sound quickly going from soft to loud – and your brain says “ow! a loud snare!” It’s such basic information that seems to be forgotten most of the time.
MFi: Many engineers tout that heavily compressed material can sound good, but its takes a lot work. I generally agree, but I am also of the belief that at a certain point, the material can never really sound great at the current industry level volumes. Where do you feel the point of no return is when it comes to applying dynamic range compression and is there some material that really warrants copious amounts of it?
Colin: The funniest thing about “loud” mastering is that it doesn’t exist. Just the simple fact that the audience sets the listening volume every time he or she listens negates the idea that there is any standard volume. You’re in your car listening to a mix CD: a quiet song comes on, you turn up your stereo to hear it better; a loud song comes on, you turn your stereo down because it’s too loud. You decide to put on a whole album instead, so you press play and then set the level to a comfortable listening level. So…why did that hyper compressed loud master make any difference?
So we’ve already established that loudness doesn’t exist because the listener always has a volume control, so what are we left with? More and less compressed. Can more compressed sound good? Of course! The point is it doesn’t matter so goddamn much. If it’s a good recording and a good mix, it will sound good heavily compressed and not compressed at all. But that said, why not use mastering to compress an amount that sounds good instead of to reach a particular VU meter average level? The current industry levels are stupid because they’re not about music or good sound, they’re about competitive marketing or more simply, getting your attention for the first two seconds of listening. But if a good recording gets a bad mastering job, it can often still sound good. Low dynamics mastering doesn’t necessarily prevent everything from sounding good at all.
MFi: Have you had a Menegroth client, either an artist or label, demand a master to be disproportionally loud in order to sound “competitive?” If so, how do you handle it? Have you had to master mixes that have been already too compressed? Again, how do you handle it?
Colin: Yes, occasionally I’ve heard this request (and from some people I never would have expected – people who should know better! Hahaha). Most of the time it doesn’t come up. Mastering is such a mysterious process for 99% of people, musicians included. They don’t know what the hell you’re doing, so they just let you do it. And yes, I get peak limited mixes all the time for mastering. When I get those, I always ask for a non-limited version. If one doesn’t exist, I just roll with it and do what I can to improve the sound (just like I do all the time – doesn’t really change my process since it’s reactionary anyway).
MFi: In general, why do you think the Loudness War exists and why do you think artists perceive louder as better?
Colin: Because loudness always sounds better for a split second. When comparing any two recordings/signals, one must always monitor them at the same perceived volume to hear differences. Human hearing isn’t linear across the frequency spectrum, so when we hear the same sound but louder, we perceive disproportionately more bass and treble, making it seem clearer and heavier. That’s the trick of loud mastering seeming better when not evaluated at a constant listening level.
If you run a record label and you don’t understand what is lost by over-compressing, you just think, “Oh it’s possible to have this mix be louder? Great! It’ll really get people’s attention on blogshmog.news.tweeter/facebook!” And if you’re a record collector and you get that new 2014 remaster of your favorite 1986 death metal album and now it’s WAY louder than your old CD, you think, “Oh cool, they turned it up! That CD was always quieter than my new CDs. They did my volume knob hand a favor!” You don’t think about how the drums don’t seem to hit hard anymore or the fact that the bass guitar is now inaudible mush because it sounds loud and shiny when you first put it on. But over time you might notice that your ears hurt after listening to the whole album and it seems to sound like shit when you crank it in the car.
MFi: There seems to be two camps regarding mastering for vinyl: One side believes that if the digital master is highly dynamic and not heavily compressed, it can be transferred to vinyl without much change. While others believe that vinyl always need a special master to compensate for the formats limitations. Which side do you stand on and how do you approach mastering for vinyl in general?
Colin: I don’t stand on either side necessarily since it depends on the source material. If your mix contains things like ultra loud out-of-phase sub bass information, it will translate to vinyl horribly. But a fairly typical rock record mixed and mastered without a ton of limiting will probably translate just fine to vinyl. Another thing to keep in mind is that the lacquer cutting engineer will often EQ and subtly process the master to adjust for the limitations of vinyl. So my M.O. is to just do the audio mastering as I normally would and let the lacquer mastering make the final compensations. One thing I notice about vinyl is that it’s not the friendliest to hyper detailed, layered mixes. So sometimes when I know vinyl will be the main format of release, I’ll exaggerate elements of the mix that are supposed to jump out.
MFi: Another hot topic is digital vs analog production. Analog proponents claim it typically yields a more pleasing product to the ears, while the digital camp believe that most of the warm and fuzzies can be simulated with the right plugins and a keen sense of EQ. Do you do the majority of work in the digital domain or the analog one or a combination of both?
Colin: Most of the work I do is all digital and all in the box. I typically do the basic tracking here at my place to 2″ tape, but I always do overdubs and mixing digitally. And now I’m in the process of getting rid of the tape machines entirely – the maintenance is too expensive and the demand for analog isn’t great enough in metal and experimental music. I’ll miss it, but at the same time I think my current all-digital recordings sound better than anything I’ve done in the past, so maybe the tape didn’t play as big a role as I thought it did. I like the process of recording on tape – so nice to be at work without a computer even ON! That’s what I’ll miss the most. I also like the limitations of not being able to do a zillion takes and edits. It encourages a more live organic approach to recording.
But as for the whole digital vs. analog “which is better?” debate: Neither. Both. Who cares? Just record music!
MFi: Have you had a Menegroth client, either an artist or label, demand a master to be disproportionally loud in order to sound “competitive?” If so, how do you handle it? Have you had to master mixes that have been already too compressed? Again, how do you handle it?
Colin: Yes, occasionally I’ve heard this request (and from some people I never would have expected – people who should know better! Hahaha). Most of the time it doesn’t come up. Mastering is such a mysterious process for 99% of people, musicians included. They don’t know what the hell you’re doing, so they just let you do it. And yes, I get peak limited mixes all the time for mastering. When I get those, I always ask for a non-limited version. If one doesn’t exist, I just roll with it and do what I can to improve the sound (just like I do all the time – doesn’t really change my process since it’s reactionary anyway).
MFi: In general, why do you think the Loudness War exists and why do you think artists perceive louder as better?
Colin: Because loudness always sounds better for a split second. When comparing any two recordings/signals, one must always monitor them at the same perceived volume to hear differences. Human hearing isn’t linear across the frequency spectrum, so when we hear the same sound but louder, we perceive disproportionately more bass and treble, making it seem clearer and heavier. That’s the trick of loud mastering seeming better when not evaluated at a constant listening level.
If you run a record label and you don’t understand what is lost by over-compressing, you just think, “Oh it’s possible to have this mix be louder? Great! It’ll really get people’s attention on blogshmog.news.tweeter/facebook!” And if you’re a record collector and you get that new 2014 remaster of your favorite 1986 death metal album and now it’s WAY louder than your old CD, you think, “Oh cool, they turned it up! That CD was always quieter than my new CDs. They did my volume knob hand a favor!” You don’t think about how the drums don’t seem to hit hard anymore or the fact that the bass guitar is now inaudible mush because it sounds loud and shiny when you first put it on. But over time you might notice that your ears hurt after listening to the whole album and it seems to sound like shit when you crank it in the car.
MFi: There seems to be two camps regarding mastering for vinyl: One side believes that if the digital master is highly dynamic and not heavily compressed, it can be transferred to vinyl without much change. While others believe that vinyl always need a special master to compensate for the formats limitations. Which side do you stand on and how do you approach mastering for vinyl in general?
Colin: I don’t stand on either side necessarily since it depends on the source material. If your mix contains things like ultra loud out-of-phase sub bass information, it will translate to vinyl horribly. But a fairly typical rock record mixed and mastered without a ton of limiting will probably translate just fine to vinyl. Another thing to keep in mind is that the lacquer cutting engineer will often EQ and subtly process the master to adjust for the limitations of vinyl. So my M.O. is to just do the audio mastering as I normally would and let the lacquer mastering make the final compensations. One thing I notice about vinyl is that it’s not the friendliest to hyper detailed, layered mixes. So sometimes when I know vinyl will be the main format of release, I’ll exaggerate elements of the mix that are supposed to jump out.
MFi: Another hot topic is digital vs analog production. Analog proponents claim it typically yields a more pleasing product to the ears, while the digital camp believe that most of the warm and fuzzies can be simulated with the right plugins and a keen sense of EQ. Do you do the majority of work in the digital domain or the analog one or a combination of both?
Colin: Most of the work I do is all digital and all in the box. I typically do the basic tracking here at my place to 2″ tape, but I always do overdubs and mixing digitally. And now I’m in the process of getting rid of the tape machines entirely – the maintenance is too expensive and the demand for analog isn’t great enough in metal and experimental music. I’ll miss it, but at the same time I think my current all-digital recordings sound better than anything I’ve done in the past, so maybe the tape didn’t play as big a role as I thought it did. I like the process of recording on tape – so nice to be at work without a computer even ON! That’s what I’ll miss the most. I also like the limitations of not being able to do a zillion takes and edits. It encourages a more live organic approach to recording.
But as for the whole digital vs. analog “which is better?” debate: Neither. Both. Who cares? Just record music!
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]]>The post Interview: Bob Ludwig Gateway Mastering Studios appeared first on Metal-Fi.com.
]]>MFi: The Stage is not only a fantastic record in its own right, but production wise, superb. And a big part of that fact has to be its clear focus on dynamics. But this isn’t A7X‘s first foray into creating dynamic metal. In fact, their last one, 2013’s Hail to the King, was also quite dynamic as well, and was the first record where you took over mastering duties. Can you give some background on how you got involved with A7X and who brought up the idea of bringing back dynamics in the first place? Was the band even aware of the Loudness War?
Bob: Much as I liked Hail To The King, I think The Stage is their tour de force with such excellent musicianship. I bet there aren’t many bands in the world who could even copy a few bars of their music, it’s so fast and precise.
The band was well aware of the Loudness War and it was part of the conversation from the get-go. A few years ago M. Shadows wrote me, “We have always been anti-loud. We need to feel the dynamics, gloss, and overall hugeness without losing the tones we worked so hard for.”
MFi: But was the band actually surprised by how much better a high dynamic master sounds over a more industry standard compressed one? At least from an outsider’s perspective, it seems like they became true believers almost overnight.
Bob: I think it is a matter of context. Some clients seem to hold Death Magnetic as the standard by which everything else is measured, and when their record has a PLR (peak-to-loudness-ratio) a few dB lower than it, they think their mix is “dynamic” in comparison. It’s not.
For Stage, I first mastered the album to be similar to King which already was much more dynamic than Death Magnetic, but the band felt that the great mixes Andy Wallace made were losing some of the detail that this fast, precise, music demanded. I mastered a version that, like the good old days, used compression only as an artistic tool to help glue some aspects of the sound together while still retaining all the transients. They were ecstatic with the result. A few days later, I almost timed it to the hour, they came back to me nervous that it wouldn’t be as impressively loud as their peers’ recordings, so I mastered a version that split the difference. They were still on the fence. I then sent them Matt Mayfield Loudness War YouTube video and shortly thereafter they decided to go with my version with full dynamics, and that’s what was eventually released. Even the radio single was not needlessly ‘hyped’ for radio as almost all A&R people think needs to be done. I was thrilled to participate in a project that proves once again you don’t need loudness to sell, you just need great music!
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MFi: And I assume the vinyl master is just as dynamic?
Bob: The vinyl is cut from the same 24-bit/96Khz mastered files as the CD, digital download, and Mastered for iTunes version. In other words, they are all equally dynamic.
MFi: Bob, you’ve won numerous Grammys for your work with practically every major artist known to man. But do you approach mastering metal differently than other genres? I mean at the very least, I would assume most metal bands come to expect a really hot master. So do you talk about the Loudness War at the onset of every new metal project you’re involved with?
Bob: Yes, most of them expect loud and bright, but not all. With loudness normalized streaming taking over bands will soon see that the louder you master the record, the more the algorithms the streaming services use will turn them down!
MFi: That’s true. But how do you balance a band’s need for volume with high fidelity?
Bob: One of the purposes of mastering is to have the clients vision transmitted to the listeners. It’s the client’s record, not mine. If they want it a certain way after sufficient investigation, then that’s exactly how it should be.
MFi: You mentioned loudness normalization. How do you feel streaming has impacted the industry with respect to volume and where do see the industry heading?
Bob: I think at this point in time a lot of bands as well as A&R people are still not educated enough about what happens with even normal radio broadcast or loudness normalized streaming, and many of them are still aren’t referencing their mixes within iTunes with Soundcheck on. When they finally become aware how the new competition has a more punchy sound rather than a squashed one, it will an enlightening experience for them!
MFi: Finally, what advice would you give metal bands going into the studio to achieve the best possible results?
Bob: Compression is the backbone of pop and metal. One can’t get the natural loud energy of pop and metal to be heard on small systems without compression. But don’t needlessly over compress just to make it loud instead of musical. Don’t forget, equalization can be somewhat reversed, but compression, once over applied can’t be backed off!
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]]>The post Meridian Audios MQA appeared first on Metal-Fi.com.
]]>Luckily, Stereophile’s resident engineer, John Atkinson, wrote a tremendously insightful article about what the technology is and a how it works. In a nutshell, MQA is not really a new format per se, (say like DSD) as it is a new form of LPCM compression that is both lossy and lossless simultaneously!
When the MQA encoder passes over a high-resolution recording (e.g. 24-bit/192kHz), it encapsulates all of the high resolution samples, in particular, frequency data about 48kHz, by overlaying them on top of itself as a standard resolution stream (16-bit/48kHz). It does this by separating each 24-bit sample into a “more significant portion” and “lower significant portion,” where the “more significant portion” represents the first 16-bits and the “lower significant portion” is the lower 8-bits. These lower 8-bits are essentially reused in the standard 16-bit/48kHz stream to encode spectrum information above the 48kHz boundary. Then depending on whether your DAC is MQA enabled or not, it will either recover just the first 16-bits of each sample, making it look like a normal CD stream, or it will extrapolate the entire 24-bit word and reconstruct the original high-resolution recording. Two for the price one!
Addendum: I believe the MQA encoder is using the lower 8-bits based on the patent link I quoted below and the claim by Meridian that MQA is backwards compatible, but I am not 100% sure. If someone can confirm, please post a comment below.
The net result according to Atkinson is a high-resolution stream that has a data rate “to be about 1.5Mbps per channel for a stereo recording, only slightly more than that required for uncompressed CD audio and about twice that required for transmission of a 16/44.1k FLAC or ALAC file (my guesstimate).” That’s pretty impressive, and makes MQA a more attractive streaming format than say pushing uncompressed PCM or even high-res FLAC.
If any of this above sounds even remotely familiar then you probably remember the whole HDCD initiative that was developed in the late ’90s and then bought by Microsoft in 2000 before they eventually abandoned it in 2005. Very briefly, HDCD encoded 20-bit sized samples in a 16-bit word thereby increasing a CD’s dynamic range. But this time around it seems MQA is more concerned about encoding frequency information than it is about extending bit-depth.
The astute reader will also note that by reducing the bit depth from 24 to 16, the 16-bit only stream is technically a lossy version of the original MQA encoded 24-bit recording. That’s right, for all of Meridian’s song and dance about how folks didn’t ask for low quality audio (i.e lossy), the fact is their MQA encoding scheme technically delivers just that. No worries though, since according to a patent filed by Meridian, that’s just peachy:
“Preferably, the first lossy representation is an accurate representation of the input audio signal other than the effects of time-invariant filtering, sample rate reduction and requantisation that imposes a time-invariant noise floor. If all quantisations, including those within the sample rate reduction, are performed to a constant bit depth and with appropriate dither, the “lossy” representation can be of a standard equivalent to CD quality and would have been considered “audiophile” reproduction only a few years ago. This is in contrast to traditional “lossy codecs” which dynamically adapt the spectral noise floor and sometimes the bandwidth in response to the input signal.”
Note, Meridian’s claim that their lossy scheme sounds better than an equivalently encoded MP3 or AAC is complete conjecture on their part. Until MQA gets peer reviewed, call me skeptical.
Now if MQA was just a clever encoding scheme I would leave it here and petition for a sample for review. Unfortunately, as I recently found out reading the official transcript of Meridian’s CES 2015 FAQ, MQA is actually just one piece to the overall Meridian puzzle:
“As will become clear in our answers, MQA is one of the technologies in Meridian’s Versatile Distribution system. MQA is also our ‘umbrella’ term for the suite of technologies and underlying audio coding philosophy, which itself goes beyond recording. Master Quality Authenticated captures the ethos of the enterprise and the MQA stream is central to distribution.
Related Pages:
The MQA technologies include: Archive extraction and creation, forensic and recording tools; advanced A/D and D/A conversion; Encapsulation to a kernel; MQA lossless coding; MQL (triply-compatible lossless bandwidth-extension) lossless compression; Playback Rendering. A glossary will follow.”
If you haven’t guessed where I’m going with this, I’ll give you a hint: It starts with ‘D’ and ends with ‘M.’
This sounds very similar to the way web-based public key certificates are issued by a certificate authority, Reading into the above a little, it seems that these keys will be used to validate a file’s pedigree and allow or block access to those precious lower 8-bits in the MQA encoded file.
Alright, perhaps it’s not DRM in the classical sense since it doesn’t actually prevent copying or playback, but it certainly seems that Meridian is shooting for having some kind of “degraded” playback mode whereby without MQA enabled software you won’t be able to play the full high-resolution stream, i.e. you’re stuck with the 16-bit “lossy” version which technically speaking is less than what you would have had with a CD. (Dual layer SACDs work the same way. You can copy and listen to the CD layer on any device, but the protected SACD layer requires a dedicated SACD player and can’t be copied. -Dave)
The idea of requiring “licensed decoders” for playback just gives me the willies. Imagine every open source player software had to pay Meridian a license fee in order to play MQA files fully. In fact, in the Atkinson article I referenced above, he even mentions that Meridian was nebulous regarding MQA’s licensing terms.
Sure it is.
I’m going to go out on a limb here but I think this is another scheme to ram DRM down the consumer’s throats, or at least has the potential to do so. As I said above, it’s not DRM in the classical sense since the MQA ecosystem doesn’t prevent unauthorized distribution or “lossy” playback. Nevertheless, it does offer content providers this degraded playback mode which maybe a tool to prevent pirating and/or legal backup. Of course hopefully I’m completely wrong and reading way too much into Meridian’s announcements. Time will tell.
Finally, it is still not even clear to me what MQA offers over plain old high-resolution FLAC other than a reduced data rate, which I admit is attractive for streaming providers like Tidal. But in the end, with folks streaming mainly over iDevices and their Android equivalents, why would anyone want to waste their data plan on higher resolution audio? Beats me.
Regardless, I am still curious about MQA and hope to get a chance to critically listen to a recording that uses it – but not Enter Sandman. Ouch.
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]]>But the intrinsic flexibility of FPGAs over DSPs also has its downsides. First, DSPs have historically been a more power conscious choice over FPGAs since their silicon can be better optimized for their specific computational workload. Second, FPGAs are simply more complex devices than DSPs. For example, let’s say you are looking to create your own custom DAC. Then the simplest route would be to pick some off-the-shelf DSP, slap it into your circuit, and feed Pig Destroyer through it. Done. With an FPGA however, you would have to go about writing all the custom code to program all of those CLBs yourself. That is no easy feat since an FPGA is typically programmed in some kind of hardware description language instead of a high-level language like C or Java. And once your code complete, you then have to suffer through this elaborate testing cycle which includes timing analysis, model simulation, and various other verification methodologies just to verify that your code actually works. So if time-to-market is critical, an FPGA based product is probably a non-starter from the get go. Finally, FPGAs almost always have higher per-unit costs because they have more transistors, and thus a bigger footprint than your comparable DSP. Remember, an FPGA vendor doesn’t know how a potential customer will use their chip, which means they typically stuff in it a lot more functionality (read: more CLBs and I/O pins) than one might need for a certain application. DSPs on the other hand, again, know exactly how they are going to be used, and can leverage that fact accordingly.
So why the heck am I telling you any of this in the first place? Because I feel that in order for you to better appreciate a product by UK based Chord Electronics, you need to first understand a little about FPGAs. As you can see, designing products around an FPGA requires a lot more expertise outside of traditional analog circuit design, which is why most of the products we are all familiar with contain off-the-shelve third-party DSPs to handle D-to-A duties. But can an FPGA based product sound just as good, or even better than a traditional dedicated DSP based design? Read on and find out.
Striking The Right Chord
My first experience with Chord was with their now discontinued DAC64 product. Now don’t hold me to the fire since it was a long time ago, but I do vaguely remember that the DAC64 had two major issues. First, it had a problem with buffering whereby there was a noticeable delay between the moment you pressed play and the metal hitting your ears. And two, and most importantly, the DAC64 was expensive (~3k), or at least seem so at the time. But despite all that, I still walked away extremely impressed since the unit looked and sounded gorgeous.
Related Pages:
Audioengine HD6 Powered Speakers
Since then, Chord has come out with a number of equally sumptuous offerings. Probably their most well known product is the Hugo, an integrated headphone amp/DAC now based around Xilinx’s top of the line 45nm Spartan-6 chipset that has built a reputation for delivering reference level sound in a somewhat portable package. But again, like all FPGA based designs, it comes at a price. The standard Hugo is $2,195 list while its bigger desktop variant, the TT, tips the scale at around the $4k mark. Not outlandish for this kind of performance by any means, but certainly not priced for your budget conscious audiophile either. Chord of course understood this, and decided to set out and design a product that would target a more younger demographic that wants great sound but on the go. It had to be truly portable as well as play nice with all manner of smart phone. And most importantly, had to have an attractive price point. So what did Chord come up with? Meet the Mojo ($599).
Meet The Mojo
First off, the Mojo is indeed a true portable device. No, it’s not dongle tiny, but at about the size of a pack of Altoids, it easily fits in the palm of your hand. But despite its relatively small stature, it features a myriad array of I/O ports, including optical TOSlink (24-bit/192kHz) and 3.5mm coaxial inputs as well as not one, but two 3.5mm single-ended headphone outputs. And those outputs offer plenty of power too, featuring 35mW at 600 ohms and 720mW at 8 ohms with an overall output impedance of 75 mOhms. In English, I was able to comfortably juice my HiFiMAN HE-1Ks as well as my Audeze LCD-3s with ease. Chord’s lead designer, Rob Watts, explained to me that he went to great lengths to preserve the small signal accuracy by using dither and noise shaping so sound quality does not vary with volume setting. Analog purists will balk, but with a product like this where IEMs and other portable headphones are going to be its chief customer, perfect volume tracking is key, and the Mojo delivers in spades.
Like its bigger sibling, Mojo’s user interface is based on a spectral polychromatic approach, i.e. status and modes are indicated through color (You mean colour. -Dave). And as you can see, each color of the rainbow indicates a different sampling rate. Same goes for volume too, which cycles through various color schemes to indicate level. If you press both volume buttons together while the unit is on, then you will cycle through a couple of different brightness settings. Hold down both buttons while simultaneously pushing the power button and that will tell the Mojo to produce a 3V line-level output. Cute.
In terms of how effective the ROY G. BIV approach to user interface is, well let’s just say I didn’t think it was nearly half as bad as John Grandberg of Part-Time Audiophile made it out to be. No offense to John, but yes, it’s very quirky, and yes, I could definitely live without all the blinky lights since it makes the unit come off a bit childish. However, after using the Mojo everyday at work, I just got used to it. I mean at the end of the day, one button turned the volume up while the other turned it down. Really, that’s about it. In terms of its sampling rate indicator, who cares? Audirvana proudly displays the rate every time I hit play. Seriously, after about one, maybe two minutes, you’ll get the hang of it easily. More of an issue for me was the fact that though the Mojo features digital volume control, you can’t control it from the keyboard. That does suck. Granted, it is by no means the end of the world, but I do think hands free operation is ideal for a device like the Mojo.
The Mojo has two mini-USB inputs, one labeled for charge and the other for digital input. Yes, you can use the Mojo while it’s charging (obviously, you need two USB cables then), but the manual points out that you are going to wait for eternity to see the charge light turn off while shooting metal through it. In general though, you get 10 hours of headbanging for 4 hours of charge. That proved more than sufficient for my daily usage. And as the color scheme shows above, the Mojo supports ever sampling rate known to man, including 768kHz DXD as well as DSD64, DSD128, and DSD512 over DoP. Basically, this little black box covers all your bases and then some. There is even some talk about a future add-on module that will plug into these ports and offer Bluetooth, WiFi, and even a SD card reader to boot. Now that would be killer.
Tap Out
A lot of Chord’s secret sauce revolves around something they call the “Watts Transient Aligned” (WTA) filter. The main thrust behind the development of this custom filter is that our brains, not ears, are really responsible for our hearing, i.e. we hear with our minds. And it turns out that according to Watts, our ears are capable of detecting a 4us timing difference between incoming sounds while the CD format, due to its limited sampling rate (44.1kHz), can only retain down to 22us of timing difference between samples. And this effects how our brain processes the sound our ears collect. Worse still, the typical interpolation filter that is at the heart of every modern DAC can not reconstruct these timing differences accurately due to their limiting processing power and thus, limited number of filter coefficients they can implement. But because Watts harnesses the processing power of a full blown FPGA, he can implement an order of magnitude more taps than a convention FIR filter, which in turn results in a more accurate D-to-A conversion with respect to transients. And so the story goes.
To accomplish all of this magic, the Mojo uses the 28nm Artix-7 chip, Xilinx’s most power efficient and lowest cost FPGA to date. Speaking of power efficiency, though the Mojo gets warm during playback, I was always able to pick up my unit after several hours of usage with ease. Even more impressive is the fact that though the Artix-7 has significantly less processing power than the Spartan-6 found in the Hugo, Watts was still able to implement his WTA filter code with very little change. In fact, the code base between the Hugo and Mojo is relatively the same, with some minor changes to the Mojo to accommodate 768kHz sampling rates. The bottom line is that for almost half the price of the Hugo, the Mojo offers the same state-of-the-art design. Nice.
Most of my listening sessions were conducted with my Jerry Harvey Audio Roxanne CIEM through my Macbook Pro since that is my daily, on the go setup. Let me get this off my chest right now: the minute I swapped out my Geek Out v1 for the Mojo, my jaw dropped. I could not believe how much better the Mojo sounded over the Geek Out. And I do mean everything sounded better – bass extension, soundstage, imaging, you name it. It was one of those moments in this great hobby of ours that you rarely have, but always live for. In fact, it sounded so great out of the box, I found myself instantly going back to records that I knew by heart just to confirm what I was hearing was indeed real. So without further ado, here are my listening notes.
Chord Electronics Mojo
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Chord Electronics Mojo
Posted by Alex on Feb 15, 2016 in Gear | 33 Comments
Introduction
Alright, I don’t expect you to actually slog through the whole video. However, if you did get through the first five minutes or so then I’m fairly confident that you have a pretty good idea of what a Field Programmable Gate Array (FPGA) is now. But to recap, an FPGA is a collection of controllable logic blocks (CLBs) that are all interconnected via a high-speed bus. In it themselves these logic blocks don’t do much. But when strung together, they can be used to perform all kinds of interesting computational work. So let’s say you want to program your FPGA to emulate a general purpose microprocessor like the Intel-based one you are using right now to read this review. You’re in luck, you can! Or perhaps you want it to perform the same functionality as an Application Specific Integrated Circuit (ASIC), say like Bitcoin mining so you can make millions of dollars by the time you actually finish reading this review? No problem. Or maybe you just want to write your own Digital Signal Processor (DSP), like an ESS SABRE, and use it in your own custom DAC? You guessed it, you can do that too. And that’s the beauty behind using an FPGA – it’s like being given your own computational Lego set that you can piece together to create whatever kind of application your heart desires.
But the intrinsic flexibility of FPGAs over DSPs also has its downsides. First, DSPs have historically been a more power conscious choice over FPGAs since their silicon can be better optimized for their specific computational workload. Second, FPGAs are simply more complex devices than DSPs. For example, let’s say you are looking to create your own custom DAC. Then the simplest route would be to pick some off-the-shelf DSP, slap it into your circuit, and feed Pig Destroyer through it. Done. With an FPGA however, you would have to go about writing all the custom code to program all of those CLBs yourself. That is no easy feat since an FPGA is typically programmed in some kind of hardware description language instead of a high-level language like C or Java. And once your code complete, you then have to suffer through this elaborate testing cycle which includes timing analysis, model simulation, and various other verification methodologies just to verify that your code actually works. So if time-to-market is critical, an FPGA based product is probably a non-starter from the get go. Finally, FPGAs almost always have higher per-unit costs because they have more transistors, and thus a bigger footprint than your comparable DSP. Remember, an FPGA vendor doesn’t know how a potential customer will use their chip, which means they typically stuff in it a lot more functionality (read: more CLBs and I/O pins) than one might need for a certain application. DSPs on the other hand, again, know exactly how they are going to be used, and can leverage that fact accordingly.
So why the heck am I telling you any of this in the first place? Because I feel that in order for you to better appreciate a product by UK based Chord Electronics, you need to first understand a little about FPGAs. As you can see, designing products around an FPGA requires a lot more expertise outside of traditional analog circuit design, which is why most of the products we are all familiar with contain off-the-shelve third-party DSPs to handle D-to-A duties. But can an FPGA based product sound just as good, or even better than a traditional dedicated DSP based design? Read on and find out.
Striking The Right Chord
My first experience with Chord was with their now discontinued DAC64 product. Now don’t hold me to the fire since it was a long time ago, but I do vaguely remember that the DAC64 had two major issues. First, it had a problem with buffering whereby there was a noticeable delay between the moment you pressed play and the metal hitting your ears. And two, and most importantly, the DAC64 was expensive (~3k), or at least seem so at the time. But despite all that, I still walked away extremely impressed since the unit looked and sounded gorgeous.
Since then, Chord has come out with a number of equally sumptuous offerings. Probably their most well known product is the Hugo, an integrated headphone amp/DAC now based around Xilinx’s top of the line 45nm Spartan-6 chipset that has built a reputation for delivering reference level sound in a somewhat portable package. But again, like all FPGA based designs, it comes at a price. The standard Hugo is $2,195 list while its bigger desktop variant, the TT, tips the scale at around the $4k mark. Not outlandish for this kind of performance by any means, but certainly not priced for your budget conscious audiophile either. Chord of course understood this, and decided to set out and design a product that would target a more younger demographic that wants great sound but on the go. It had to be truly portable as well as play nice with all manner of smart phone. And most importantly, had to have an attractive price point. So what did Chord come up with? Meet the Mojo ($599).
Meet The Mojo
First off, the Mojo is indeed a true portable device. No, it’s not dongle tiny, but at about the size of a pack of Altoids, it easily fits in the palm of your hand. But despite its relatively small stature, it features a myriad array of I/O ports, including optical TOSlink (24-bit/192kHz) and 3.5mm coaxial inputs as well as not one, but two 3.5mm single-ended headphone outputs. And those outputs offer plenty of power too, featuring 35mW at 600 ohms and 720mW at 8 ohms with an overall output impedance of 75 mOhms. In English, I was able to comfortably juice my HiFiMAN HE-1Ks as well as my Audeze LCD-3s with ease. Chord’s lead designer, Rob Watts, explained to me that he went to great lengths to preserve the small signal accuracy by using dither and noise shaping so sound quality does not vary with volume setting. Analog purists will balk, but with a product like this where IEMs and other portable headphones are going to be its chief customer, perfect volume tracking is key, and the Mojo delivers in spades.
Like its bigger sibling, Mojo’s user interface is based on a spectral polychromatic approach, i.e. status and modes are indicated through color (You mean colour. -Dave). And as you can see, each color of the rainbow indicates a different sampling rate. Same goes for volume too, which cycles through various color schemes to indicate level. If you press both volume buttons together while the unit is on, then you will cycle through a couple of different brightness settings. Hold down both buttons while simultaneously pushing the power button and that will tell the Mojo to produce a 3V line-level output. Cute.
In terms of how effective the ROY G. BIV approach to user interface is, well let’s just say I didn’t think it was nearly half as bad as John Grandberg of Part-Time Audiophile made it out to be. No offense to John, but yes, it’s very quirky, and yes, I could definitely live without all the blinky lights since it makes the unit come off a bit childish. However, after using the Mojo everyday at work, I just got used to it. I mean at the end of the day, one button turned the volume up while the other turned it down. Really, that’s about it. In terms of its sampling rate indicator, who cares? Audirvana proudly displays the rate every time I hit play. Seriously, after about one, maybe two minutes, you’ll get the hang of it easily. More of an issue for me was the fact that though the Mojo features digital volume control, you can’t control it from the keyboard. That does suck. Granted, it is by no means the end of the world, but I do think hands free operation is ideal for a device like the Mojo.
The Mojo has two mini-USB inputs, one labeled for charge and the other for digital input. Yes, you can use the Mojo while it’s charging (obviously, you need two USB cables then), but the manual points out that you are going to wait for eternity to see the charge light turn off while shooting metal through it. In general though, you get 10 hours of headbanging for 4 hours of charge. That proved more than sufficient for my daily usage. And as the color scheme shows above, the Mojo supports ever sampling rate known to man, including 768kHz DXD as well as DSD64, DSD128, and DSD512 over DoP. Basically, this little black box covers all your bases and then some. There is even some talk about a future add-on module that will plug into these ports and offer Bluetooth, WiFi, and even a SD card reader to boot. Now that would be killer.
Tap Out
A lot of Chord’s secret sauce revolves around something they call the “Watts Transient Aligned” (WTA) filter. The main thrust behind the development of this custom filter is that our brains, not ears, are really responsible for our hearing, i.e. we hear with our minds. And it turns out that according to Watts, our ears are capable of detecting a 4us timing difference between incoming sounds while the CD format, due to its limited sampling rate (44.1kHz), can only retain down to 22us of timing difference between samples. And this effects how our brain processes the sound our ears collect. Worse still, the typical interpolation filter that is at the heart of every modern DAC can not reconstruct these timing differences accurately due to their limiting processing power and thus, limited number of filter coefficients they can implement. But because Watts harnesses the processing power of a full blown FPGA, he can implement an order of magnitude more taps than a convention FIR filter, which in turn results in a more accurate D-to-A conversion with respect to transients. And so the story goes.
To accomplish all of this magic, the Mojo uses the 28nm Artix-7 chip, Xilinx’s most power efficient and lowest cost FPGA to date. Speaking of power efficiency, though the Mojo gets warm during playback, I was always able to pick up my unit after several hours of usage with ease. Even more impressive is the fact that though the Artix-7 has significantly less processing power than the Spartan-6 found in the Hugo, Watts was still able to implement his WTA filter code with very little change. In fact, the code base between the Hugo and Mojo is relatively the same, with some minor changes to the Mojo to accommodate 768kHz sampling rates. The bottom line is that for almost half the price of the Hugo, the Mojo offers the same state-of-the-art design. Nice.
Time To Get My Mojo On!
Most of my listening sessions were conducted with my Jerry Harvey Audio Roxanne CIEM through my Macbook Pro since that is my daily, on the go setup. Let me get this off my chest right now: the minute I swapped out my Geek Out v1 for the Mojo, my jaw dropped. I could not believe how much better the Mojo sounded over the Geek Out. And I do mean everything sounded better – bass extension, soundstage, imaging, you name it. It was one of those moments in this great hobby of ours that you rarely have, but always live for. In fact, it sounded so great out of the box, I found myself instantly going back to records that I knew by heart just to confirm what I was hearing was indeed real. So without further ado, here are my listening notes.
I’m a huge Rammstein fan. YUGE! On November 12, 2010, after 10 years of not setting foot on US soil, Rammstein played to a sold out audience of over 18k people at Madison Square Garden in New York City. I was there. It was incredible. I’m still trying to find myself on the Blu-ray disc they just released of the show! In addition to this live concert set, these crazy Germans also released the mother of all vinyl boxsets that contains their complete discography remastered for black 180g vinyl. My wife bought me this boxset for my Christmas present (she’s clearly a keeper) and I have been enjoying it ever since.
Like so may other US fans, I became a fan of the band when I first heard “Du Hast” on the radio off of 1997’s Sehnsucht. Their magnum opus however is probably that record’s follow up, 2001’s Mutter, which of course I could hum in my sleep at this point. Listening to a needle drop of it on the Mojo was really something magical. Not only are the dedicated vinyl masters an order of magnitude better than their crushed CD counterparts, but sound absolutely glorious on the Mojo. John Darko over on DAR calls the Mojo’s sound piquant. I could not agree more, though on this side of the globe I would of said alive. Everything sounds and feels sonically more alive through the Mojo. Listening to “Ich Will,” it was impossible not to ride the cymbals with my head, as their reverb sounded airy and real. No muddiness, no sense of awkward treble decay – they just sounded natural. Same is true for the low-end as well, as bass extension is equally ungodly. I really believe that this might be the first time that all of my Roxanne’s 12 drivers were actually being put through the ringer! Call me impressed.
As another test, I threw my Roxanne’s into iFi Audio’s iDSD for a little ABX comparo (I tried my darnest to level match as best I could). With the Roxanne’s at least, I preferred the Mojo almost every time. They both seem to be able to eek out all the little sonic nuances these vinyl masters have to offer, but the Mojo’s airy, spacious presentation just put it over the top.
Staying with our wax motif, I decided to bring out an oldie but goodie, and one that I’ve used in the past as review source material, a needle drop of Bolt Thrower‘s 2005 epic Those Once Loyal. It is still my favorite BT record to date and one that continues to get heavy rotation. But for this listening session I decided to switch ears and plug in my HE-1K instead for some more Mojo/iDSD ABX fun. Detail retrieval was still top notch through both, with tracks like “The Killchain” and “Anti-Tank (Dead Armor)” roaring out the gate in piquant fashion. What I found most fascinating however, is that though the iDSD seemed to eek out more mid-bass thump, it was clearly at the sacrifice of clarity. For example, on the track “Entrenched,” the immediate groove filled buzz of Ward and Thompson’s guitars sounded ostensibly clearer through the Mojo. Same was true with bass, as Jo’s presence seemed more integrated into the whole on every track. Don’t get me wrong, the iDSD was no slouch either, and with its robust power supply, easily kept up. But through the Mojo, the music sounded bigger, bolder, and again, just more alive.
As my in-depth review of Dystopia revealed, I think this is the best record Sgt. Mustaine’s Lonely Hearts Club Band have released since Youth. Regardless, one of the unfortunate aspects of Megadeth‘s latest offering is that its production is quite poor, with the whole album pushed to near inches of its sonic life. I am still hoping for a FDR version to be released like the last two, but so far, no dice. Ergo, the album is excellent fodder for the Mojo to gauge how it handles overly compressed material.
Truth be told, I was a bit fearful that Mojo’s infatuation with accuracy and wide soundstage might actually prove to be to its detriment when confronted with low DR material. Let’s just say my worries were put to rest rather quickly, as the Mojo sounded remarkably civil despite the fact that the recording I was pumping through it was not. Sure, while listening to “Post-American World” or “Conquer or Die,” I cringed at the muddiness of the guitars and the complete lifelessness of the drums. But that was clearly not the Mojo’s fault, and in fact, I thought it worked its magic as as much as it could, pulling out Ellefson’s bass out of production oblivion as well as giving Adler a little bit more oophm where say the Geek Out would simply sound competent. The Mojo time and time again offered big sound regardless of the source material’s production value.
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]]>The post Audioengine HD6 Powered Speakers appeared first on Metal-Fi.com.
]]>Anyway, as I said, no floor standing speakers, so I’m thinking bookshelf sized. And even though I’ll probably mount them on some snazzy looking speaker stands, for the time being they are going to have to be strategically placed on my mantle. So whatever I do get needs to be somewhat forgiving with respect to room acoustics. I also need to be able to stream music through them since my wife can easily replace me but not her Spotify playlists. Finally, I don’t really have room for a dedicated amp and/or DAC right now, which means any pair of speakers I do decide on has to be either self powered or require a Raspberry Pi sized amp to juice them.
Does any of the above sound even remotely familiar to you? I’m going to go out on a limb and say that it does, or some flavor of it. If so, step into my office, and meet the HD6.
Start Your Audioengine
I think at this point anyone who calls them self an audiophile knows of, or at least has heard of Audioengine. But in case you have been living in a True Norwegian igloo for the past decade, Audioengine was founded in 2002 by a team of audio professionals. Their first prototype, a pair of studio monitors, was originally meant for the professional studio market; until the team came to a realization that with a little tweaking these new monitors could easily be adopted for general purpose use. Fast forward to 2005, where lead engineer and co-founder Dave Evans discovered that his new babies also sounded good playing lossy AAC files over the air via Apple’s then relatively new Airport Express base station. He then had an epiphany: if he modified the enclosure slightly he could retro fit a built-in amplifier and basically create an entire stereo system in one tidy little package. Soon after, the A5 was born.
Suffice it to say, it was a smashing success. How could it not be? It was sleek, affordable, sounded better than most of its contemporaries, but most importantly, it embraced the digital revolution on day one. And by “embrace” I mean that the A5 and its follow-up, the A5+, were products that were geared for audiophiles who have no qualms whatsoever using their iPhone or other smart device as a source, which is why both products featured a USB charge port. And ever since, Audioengine has continued to make great sounding products that are affordable yet modern, offering features that give audiophiles what they really want but afraid to actually admit it in public. Trust me on this one.
But though Audioengine has always built great sounding products, they’ve never really built high-end ones. Well, perhaps that’s not completely fair – their A2 computer speakers are certainly “high-end” but they still don’t compare to a real, honest-to-goodness reference stereo system. Putting it another way, a lot of audiophiles may begin their journey with Audioengine but don’t necessarily finish it with them. But in 2015, that all changed with the introduction of the HD6, Audioengine’s new flagship bookshelf speaker system. And the price? $749 buckaroos. Hold on a minute. That must be a typo. Let me double check their website. Stand by. Flagship. $749. Flagship? $749? Flagship! $749! Hot damn.
The Setup
When I finally received my review sample (thanks Brady!) my first reaction as I was taking them out of the most well-designed, well-packed shipping box ever conceived by man was, “Holy cow! These little beasties are heavy!” The right passive speaker weighs in at 12.5 lbs. while the active left side that has the amplifier and other circuitry in it clocks in at whopping 17.5 lbs.! Heavy metal indeed. And as I soon discovered, their impressive girth also has a lot to do with the enclosures themselves which are made of thick medium dense fiber that comes in a walnut, cherry, and black satin finish. If you are on the fence of which one to buy, don’t be. Get it in a wood finish. I assure these pictures do not do the HD6 justice. The walnut finish looks gorgeous.
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Moving to the front of the speaker, the left one has a white power indicator light as well as the volume control knob. Both speakers come with a magnetically attached black mesh grill that can be easily detached for a more pro-studio look. I actually prefer the grill on as a protective measure for daily use and the “naked” look when I want to impress friends and family. Either way, you can’t go wrong. Oh, and if manual knob rolling isn’t your thing, no problem, Audioengine ships a really nice solid aluminum brushed remote at no extra cost.
Starting from the top of the left active speaker, you have your Bluetooth antenna and pair button, 3.5mm coax and optical inputs, dual RCA L/R inputs as well as a variable output, speaker binding posts to connect to the right speaker, power switch, voltage select, fuse, and last but not least, the power inlet. Speaking of power, the transformers used in the HD6 are gapless core toroidal, which is cooler and lighter than your typical transformer but without any loss of power. That also translates into better bass response and more efficient operation.
Now the absolute easiest way to get these puppies going is to connect the left speaker to the right with the included 13ft 16AWG speaker cable, select your country’s appropriate voltage setting (the HD6s are US ready right out of the box), plugin the power cord and then hit the switch. The front white power indicator will then turn on and the speaker will go directly into pairing mode (the Bluetooth LED light on the back will start blinking). Switch back to your desktop or other Bluetooth capable device and you should be able to pair with the HD6. Note, the HD6 will remember up to six different devices so you only really have to pair once. Awesome. If you aren’t so keen on wireless operation however, then use the provided mini-coax or RCA cables and hook the speaker’s analog inputs directly to your favorite source. I was in up and headbanging in about two minutes flat.
Driving The HD6
The HD6, at least spiritually, is really no different than any of the other speakers Audioengine offers in that it is another two-way design. But how Team A went about implementing it is quite another story entirely. The newly designed 5.5″ woofers are made from a Kevlar woven glass aramid composite and are set in with rubber surrounds. By using Kevlar, which is a very strong material, each woofer is able to retain its shape despite having a much larger excursion pattern. That equates to tighter and deeper bass response (Audioengine claims the HD6 has no problem hitting the 50Hz mark with ease!). On the other side of things, the 1″ tweeters are a ferrofluid-cooled silk dome type that uses neodymium magnets. The astute reader will note that this type of design is fairly common in a lot of audiophile grade reference designs, but Dave Evans and Co. have meticulously tuned each driver so that under heavy load they still exhibit a smooth response.
The built-in amplifier is a class A/B type capable of a whopping 150 watts of peak power, which equates to 50 watts RMS or 75 watts peak per channel, i.e. these puppies get pretty loud. THD+N is less than 0.05% across all power settings with a respectable >95dB SNR (A-weighted). Clearly, Audioengine went out of their way to ensure that the HD6 looks and at least on the surface, measures well.
On the digital side of things, Audioengine uses the AKM4396 chipset, an industry stalwart and capable of handling sampling rates up to 24-bit/192kHz. Why the 4396? Well first, the folks over at Audioengine have a lot of history with this particular chipset, as it is at the heart of their well-received D1 portable DAC. Next, the 4396 has built itself a solid reputation over the years for its clean sound and has been vetted by companies like Schiit and CEntrance. Now I’ll be honest with you, the 4396 has never wowed me. However, I’ll be the first one to admit that it’s a solid chipset none the less and certainly a fine choice for the HD6.
Racer aptX
Historically speaking, audiophiles don’t pair well with Bluetooth (Drum roll, please. -Dave). And a lot of that has to do with the fact that Bluetooth as a technology was never really meant for high fidelity playback. So that begs the question, why would Audioengine include it then into their top of the line speaker system?
Let’s step back for a moment and understand a little about Bluetooth and its relationship with streaming. Since it’s inception, Bluetooth has always been about devices wirelessly communicating with each other over short distances. And in order to do that, the Bluetooth spec defines a common set of protocols that every device must adhere to in order to earn that Bluetooth compliant logo. These protocols are further broken down into functional units called profiles. For a device to support streaming audio over Bluetooth, it must implement the Advanced Audio Distribution Profile (A2DP), which states that if a device wants to stream audio it must be able to handle the Low Complexity Subband Coding (SBC) codec and can optionally support other codecs such as MP2, MP3, AAC, and of course good ‘ol ATRAC (Viva la Minidisc! -Dave).
Notice something? All the codecs are lossy, and that’s by design since Bluetooth as a protocol is very bandwidth limited. Worse still is the fact that out of all the codecs listed above, only SBC is mandatory, which has a really low bitrate. So it’s no wonder why the first time audiophiles heard Moonsorrow over Bluetooth it sounded like hot garbage.
However, in their wonderful write up, Audioengine outlines why the company is a big believer in Bluetooth as a streaming technology and more importantly, why their implementation of it addresses a lot of the concerns I outlined above. First, in the HD6, Audioengine implementation supports AAC natively. So if you are buying all of your music off of iTunes, you’re covered. For the rest of us sane people however, it also includes native aptX support, a codec which has gotten a lot of attention over the past few years as a third-party, higher fidelity alternative to SBC. That’s good news too since most Android devices default to using aptX when it’s available for streaming. I even verified that my Macbook supports aptX as per Darko’s great article found here. If you’re curious whether or not your device supports it, check the official list to confirm. Finally, I experienced practically no drops outs, pops, poops, you name it while streaming over Bluetooth. The range of the HD6’s receiver is just downright incredible, and you will be hard pressed to loose connectivity during normal operation. Bottom line: if you are on the fence about streaming directly to the HD6, don’t be. I’m fairly confident that you won’t walk away completely blue in the face. Or maybe you will?
Pick A Room, Any Room
If you go on the Audioengine website right now, you’ll note that not only do they claim the HD6 is their best sounding speaker to date, but also their most versatile. And though that probably has a lot more to do with the sheer number of ways you can connect the HD6 to your favorite source, I also got to believe it partially has to do with the fact that these speakers sound pretty much great any where you put them. So far I have deployed the HD6 in my living room, on a pair of pseudo-speaker stands, as well as even tried them as desktop speakers with no really issues to speak of.
And that brings me to another interesting point about these little beauties, their near field performance is excellent, and if your desk is indeed big enough, I can whole heartily recommend the HD6 as an all out desktop speaker replacement. I realize at $749 that sounds ridiculous (Or maybe I just roll that way! -Dave), but I just want to instill in you that these speakers are very versatile. Obviously, my favorite setup was in the living room since that is their natural habitat (read: my wife really likes them so any attempt I made to hide them from her for this review was met with heavy resistance). For the most part though, these “walnut wonders” were able to image properly without too much fussing. There were of course a few times where it was obvious my placement choices were so bad that even the HD6 revolted. But the bottom line is that they aren’t very directional, and I was easily able to get them to sound good to great in almost every room I tried them in.
Listening Notes
Short story: the HD6 did not disappoint.
Long story: When I evaluate any kind of serious stereo setup, I’m not looking for just clinical accuracy. Save that for the studio. What I want from a stereo is a setup that is involving and actively engages with its audience. And by ‘engage,” I mean a system where when I hear music playing through it, I immediately want to sit down and truly absorb the sound waves hitting me. Oh, and by “sit down” I mean air guitar – lots and lots of air guitar.
I’ve been on a massive Darkthrone kick of late, literally listening to their entire back catalog starting from their 1992 classic, A Blaze in the Northern Sky, on up (I’ll get to Soulside Journey eventually). Obviously, their early records aren’t exactly stellar examples of high fidelity, but no matter. They will always hold a special place in my black heart. And to this day, Blaze is still an absolute bona fide classic, and to my ears, still holds up against all the wannabees and copy cats that have come after it. The tag team of Nocturno and Fenriz really broke new ground when they unleashed this beast on the world, and metal has never been the same ever since. This is True Norwegian Black Metal at its finest.
How did the HD6 do? Really well, almost surprisingly so. The first aspect of the HD6 that was immediately apparent is that as I mentioned above, they image well. The musical picture they present to your ears and brain is quite coherent, giving you at true sense of the album’s mix (or in Darkthrone‘s case lack thereof). Transients were also excellent too out of the box, since I found the overall experience to be highly energetic – Fenriz really sounds like he banging away smack dab in the center of my living room. Horns up!
With respect to 20-20, bass was always present, sounded deep, but not as tight as it could be, which comes to me as no surprise given the size of these speakers. I did try to weigh them down a bit with some heavy hardcover books which made a slight improvement, both in the bass and soundstage departments. This came to me as no shock since given their bookshelf size, they can only do so much. The highlight of the show by far was the mids, which probably has some emphasis in the mid-bass region that gives them a very warm sound, almost tube like. Put simply, Nocturno’s buzzsaw was literally dripping off the HD6, and I was loving every minute of it.
Remember I said I had to share this review with my wife? Well, that meant some form of sonic compromise. So I decided to pick something for her as a way to prove my worth as a husband (For now at least. -Dave). If you ever find yourself in this same exact boat then may I recommend Austin Wintory‘s incredible soundtrack to the game Journey. It is in my humble opinion a sure fire crowdpleaser. For this test I decided to hook the iDSD’s line-level output to the HD6 directly and use it as a source instead of the built-in AKM. The result? Sublime. Again, the 4396 is a fine chipset, but the iDSD’s innovative bit fondling is a lot better, and the HD6 rewarded me with a much richer, fuller sound. Tina Guo’s cello was just jaw-dropping beautiful. Same goes for Amy Tatum’s flute, which had a new level of clarity through the iDSD. What all of this tells me is that there is room for growth here which is exactly what you want out of a pair of reference level speakers.
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