APPLE MAKES NEEDED CHANGE TO SOUND CHECK IN ITUNES

Sound Check is a very convenient feature in iTunes that normalizes the playback volume across all songs, so you will not have a song that was brickwall mastered blasting out 5 times louder than a song that was mastered with more dynamics intact. This is extremely useful if you have created playlists that include songs from different eras and genres that have widely varying volumes, or if you are just playing your library in shuffle mode.
Until now, a problem with Sound Check occurred if you were listening to a complete album in album mode. Artists/producers often purposely mix some songs on an album at lower volumes to create an emotional effect through dynamics, or volume differences (that's why dynamics are referred to as "expression" in music, the dynamics express emotion). And if the album includes both electric and acoustic songs, the acoustic songs should be perceptually lower in volume or they will diminish the  electric tracks. But with Sound Check turned on, it adjusted the level of the all the songs to a common volume, destroying the effect the artist/producer intended.
The good news is in an iTunes update sometime around the end of July, Apple made an unannounced change to Sound Check. Now, if you are playing iTunes in album mode with Sound Check, the original volume differences between tracks on the album will remain intact, with the overall volume of the album being brought down so the loudest song is at the AES recommended volume for streaming.
It's not perfect yet; it doesn't seem to read the volume of the loudest song properly, so if you switch from song or playlist mode - where Sound Check is adjusting the volume of each song - to album mode, the album mode is louder than song/playlist mode. Not up to full volume as without Sound Check, but just enough to be a little annoying and require a volume adjustment. But that's a very minor detail that should be ironed out soon, the point is the dynamic flow of the album is preserved without having to go into preferences and turning Sound Check off.
If you're not familiar with Sound Check,  open iTunes, go to the top left "iTunes" pulldown menu and select "Preferences". At the top of the preferences window select the "Playback" tab. You'll see a checkbox to turn Sound Check on. Also, be sure "Sound Enhancer" is turned off. And, if you want to hear the album exactly as intended, turn "Crossfade Songs" off as well.

SPOTIFY FINALLY LOWERS ITS VOLUME! THE LOUDNESS WARS ARE PRACTICALLY FINISHED!

Sometime around May 20th, Spotify finally lowered its volume normalization to approximately -14LUfs, -3dB lower than their previous level and generally matching the levels of TIDAL and YouTube. This is big news. Spotify was the last holdout, extending the loudness wars by being 3dB louder than other streaming services,  pressuring musicians, labels and mastering engineers to make needlessly hot masters to compete loudness-wise, even if it didn't suit the style of music. With this move Spotify, TIDAL and YouTube now use playback reference levels within a dB of each other, with Pandora, ReplayGain and Apple Sound Check being -2dB lower at -16LUfs, exactly conforming to the recommendations of the Audio Engineering Society for streaming loudness.

This means there is no longer any pressure to make hotter masters to compete volume-wise on Spotify, or any other streaming service; you can now have your mastering engineer create masters at an average integrated loudness of -14LUfs and it will work perfectly across all streaming services. (YouTube is actually +1dB hotter at -13LUfs, so if max loudness on YouTube is important to you, you might push everything to -13LUfs, knowing that Spotify, Tidal and others will turn it down a dB. But if your music is fairly dynamic that extra dB really won't make a difference, and 1dB less limiting will certainly make a difference!)

Even though the -14LUfs level is still 2dB hotter than the AES recommendation, this is a HUGE step forward, allowing an extra 3dB of peak-to-average headroom compared to Spotify's previous level, and a whopping 6dB improvement over the most ridiculously loud CD's.

The Good News and the Bad News
The good news is that with the death of CD and now with the willingness of streaming services and media players to comply, at least within a couple dB, with the loudness guidelines set by the AES, there is now no need to push volumes on your production in order to be competitive with other music. The result will be much more open, punchy, expressive music, and music more pleasant to listen to without all of the distortion and artifacts caused by hyper-limiting. These new levels are essentially the same as when music was mastered for vinyl, and that's one of the reasons people love the sound of vinyl from the 50's through the early 90's. It's partly the analog sound of the record and the cartridge, but a huge part of it is that the clarity, emotion and punch of the music was not destroyed by being over mastered.

The bad news is, all of the music that has been ruined since the first stupid-loud CD came out in 1995, and everyone else jumped off the cliff right behind it to be competitive in the loudness wars, will still sound just as awful compared to newer, more dynamic productions. All that will happen to it is it will be turned down to the normalized loudness of the steaming service or media player, so it will no longer be louder than other songs, but all the damage done to that music to make it loud will still be there.

For over 22 years, the answer to "Why does that music pop when it comes up on the CD changer?" has been "Because it's louder than the others". Now the answer to the new question of "Why does that music pop on Spotify/TIDAL/iTunes?" is "Because it's more dynamic than the others". This is a great day!

There is one group that will be big losers in this, and that is the plethora of people claiming to be "mastering engineers", along with mix engineers who tell you they can also master, but who know nothing of mastering other than to make it loud. The art of mastering, the actual sweetening process, will once again be what mastering is all about. Does the person mastering your music even know what Integrated, Max Momentary and Max Short Term loudness is? Does he have the meters to measure this? Does he understand the important interaction of those? Does he know what True Peak means, and does he even have a True Peak limiter? Does he know how to master at proper True Peak levels so data-compressed formats don't distort? Does he understand crest factors so that your music does not needlessly get turned down on streaming services even though your integrated loudness is correct? At DES Mastering we have been on top of this since around 2011 when the new loudness measuring methods were being developed, and attended the original AES seminars and workshops that helped established the new standards for broadcast and streaming loudness. And we have the tools and the knowledge to create masters for you that will take full advantage of this breakthrough in sound quality!

CONTROL YOUR SIBILANCE AND PLOSIVES!!!

Sibilance (plural 'sibilants') is the piercing high frequency 'hiss', if you will, created mostly by s and t sounds in vocals.  (Sibilants normally live somewhere around 7kHz.) There has to be a certain amount of sibilance for it to sound like an s or t, but everyone's voice has different amounts. For instance, my wife has a friend whose voice produces such extreme sibilance I can't stand to listen to her talk, it absolutely pierces my ears. Plosives are on the other end of the spectrum; they are the 'booms' created by aspiration, which is the forceful release of breath on certain words, hitting the diaphragm of the microphone.

Sibilants and plosives can sound bad enough in your mix, but they will be highly exaggerated in the mastering process, especially due to the excessive limiting most people demand in order for their music to be loud. And Lord knows, being loud is what's important these days above all else. Yes, the mastering engineer can help these to some degree, but the mastering lab is not the proper place for the these to be addressed. The proper place is while the vocals are being cut. Of course the tracking engineer can resort to de-essers and notch filters, but before any of that is used they should first work with mic selection. Many condenser microphones, which is what most recordists will grab first for vocals, have a certain amount of a presence peak. Sometimes this is due to the nature of the design of the mic, but often it's a trick being played by the manufacturer, especially in cheap condensers, to give the the false impression that the mic has a wonderful 'air' and brilliance, when instead it's just an exaggerated top end that you could simulate with eq. So if you use one of these mics on a vocalist who already has exaggerated sibilance in their voice, you have a mess on your hands. The lazy recordist, if they're aware of the problem, will just throw a de-esser and/or notch filter on the track and be done with it. But as i've said before, you should strive to use as little processing as possible in every step of your recording and mixing. So taking a little time to get up and down and trying different mics to find the one that at least doesn't exaggerate the problem is the proper thing to do. Ribbon mics most often have a creamier, smoother top end than condensers, and also don't ignore quality dynamics like the Shure SM-7 (used on many or most of Michael Jackson's vocals, for instance) or the Sennheiser 441 (used on TomPetty and Stevie Nicks, for instance). Then even if you do still have to use the processors, you're using them less. 

The plosives are caused, as I said, by aspiration hitting the diaphragm of the mic, so that is a simpler matter to fix by just working with the placement of the mic first so the breath blast doesn't hit the diaphragm, and then using the good ol' pantyhose pop filter if that doesn't completely solve it. 

If the problems didn't get addressed during tracking, the next preferable fix would be during the mix. If it is just pure sibilance, and you are working with a ProTools type digital recording system, it's best to take the time to find the sibilant in the vocal track, make an edit on either side of it, and then lower its volume. But the problem is that sibilants have such extreme energy that in some cases they will also cause a splatter of high frequency distortion from either overdriving the mic preamp or the a/d converter or both. In that case you have a much longer duration splat than than the actual sibilant itself was, and you'll have to resort to some de-essing or 'spot' eq'ing. 

If you are working with a vocalist who has extreme sibilance (I certainly hope my wife's friend doesn't sing or do voice work. If so, I pity the poor engineer), then the best thing to do (after mic selection, which won't solve all of the problem in these cases, but will help) is to add a little de-essing and/or filtering while you're cutting the vocal, and then a little more while you're mixing. This will be MUCH more transparent than trying to fix it all in one step or the other. 

As I said at the top, your mastering engineer will have de-essers and filters to add to the mix to at least help control these problems, but that's the worst solution, as he will not be applying these to just the vocal but instead across the entire mix. And if the vocal has had a considerable amount of reverb added to it, the mastering engineer will be able to kill some of the initial attack of the sibilant or plosive, but the trail in the reverb will still be there, which might draw even more attention to problem than just leaving it alone. If it had been properly addressed during tracking and mixing, the reverb trails will be clean. 

Mastered for iTunes at DES Mastering

There are over 250 million iOS devices in the world today connected to Apple’s vast iTunes ecommerce system. Considering the ubiquity of such a digital audio-ready footprint, there’s a fairly strong chance that just about any commercially-released music title will, at some point, be heard on an iPod, iPhone, or iPad. The resulting market of music consumers who want their content with them to enjoy on-the-go, at home or anywhere dictates that professional mastering engineers should plan and monitor the output of their efforts to provide the very best listening experience possible on iOS devices.

Mastered For iTunes - What is it?

During the first quarter of 2012, Apple implemented their new high-resolution encoding process for creating AAC's. (An AAC is what you download from iTunes, and is the iTunes version of an mp3, both being data-compressed audio files.) In order to take full advantage of the improved quality of this new encoding, Apple has released a set of guidelines and tools to assist mastering professionals in delivering optimized source files for this new high-resolution encoding and ecommerce process, allowing the mastering engineer to audition the changes that take place during AAC encoding prior to committing your mix to processing. Apples’s goal is to build awareness of best practices that will help to ensure that consumers have a great listening experience on their mobile hardware. When these guidelines are followed, the music is considered to be "Mastered for iTunes". DES Mastering supports Apple's efforts and has the full set of tools and the knowledge to prepare finished masters for release into the iTunes environment with the best possible quality. 

It All Starts with Tracking and Mixing in High-Resolution

If you have been following our blog, you know that I have emphasized over and over the importance of tracking, mixing and mastering in high-resolution, meaning 24 bit, and preferably 96k, but at the very least at 48k/24 bit. Apple is now requesting that their music partners submit their source files at 96k/24 bit as the ideal in order to take full advantage of their new encoding process.

As we have pointed out in previous posts, this does not mean that you should sample rate convert your 44.1k/16 bit files to 96k/24 bit. This will do nothing for you. Even if you use the best mastering-quality SRC, the resulting 96k file will sound exactly like the 44.1k file, and from most commercially available SRC's the 96k file will actually sound worse than the original 44.1k file due to artifacts and distortions created during the sample rate conversion. 

So once again, here are the important points:

If possible, track and mix at 96k/24 bit. At the very least track and mix at 48k/24 bit. 

If you are mixing 'in the box', do not perform any sample rate conversion. The mix files should be the same sample rate that the project was tracked at. Again, upsampling a 44.1k or 48k mix to 96k will gain nothing, and will potentially lose quality. 

Do as little processing to the overall mix as absolutely possible, and preferably none. This means little or no eq, compression or limiting applied to the stereo master buss. This doesn't mean that you should not eq or compress individual tracks as you see fit - just try to leave the stereo buss clean of any processing. 

Leave headroom in your mix. if you are doing a proper 24 bit mix, there is no reason for the hottest peaks in your mix to exceed -3dBfs, and ideally they will sit at -6dBfs. 

For more detailed info on this see Mixing in Preparation for Mastering Part 1 and Part 2.
Have a professional master your project. Whether you use DES Mastering or not, you should have a mastering professional finish your project. You may have the best mix engineer in the world, but that does not qualify him as a mastering engineer. It is a completely different skill set, and a completely different way of using seemingly familiar tools. Your project should be mastered in a proper mastering room, not a control room, by a mastering engineer, and preferably one who knows how to create the best possible mastered files for downloadable formats. 

The Best Encode Begins with DES and soundBlade

Encoding for any lossy format such as AAC or mp3 benefits from the highest quality source file possible. This is because the end result of the encoding process is a data compressed file that affects the audio quality. To achieve the best encoded result, it is important to start with the highest quality sound possible, such as that derived by DES Mastering using the Sonic soundBlade mastering system. The Sonic Studio Engine (SSE) used in the soundBlade system enables key processing to deliver pristine sound that is free of artifacts that can be caused by other lesser audio engines. Just because a manufacturer works "Pro" into the name of their product does not make it a mastering-quality platform. When Apple creates an AAC from a soundBlade master it will yield the most desirable output suitable for publishing on the vast iTunes ecommerce platform. 

Remastering for iTunes

If you have a back-catalog of music that is already available in the iTunes Store, or that has not yet been released on iTunes, you might consider having DES properly re-master your project for iTunes delivery. This would require you to have your original mix files, and those will hopefully be 24 bit and 48k or above. If you submit your remastered music to Apple, they will perform a series of tests, and if they determine that the proper guidelines have been followed and that there is an improvement over your previous release, this will earn you the "Mastered for iTunes" badge. 

If you mastered with DES at any time over the past 3 years, we more than likely still have your project in our archive, and if you delivered mixes in high-resolution, then it will be a relatively simple process to create mastered high-res files as Apple is now requesting; it would not require a complete remastering. Contact us for details. 

Importing Your Hi-Res Audio to the iPhone/iPad

Again with the hi-fi related post! But if you are in music production, I promise the training of your ear is just as important as any knowledge you have of audio gear and recording techniques. And having a quality system to listen to music on outside of the studio is important.
If you're following my posts detailing how to get audiophile-quality reproduction out of your computer using standard iTunes and other media players, in Part 1 you learned that the iPhone/iPod can play AIF and WAV files at 44.1k/16 bit CD-quality and ALAC files at 44.1k/16 as well as 24 bit. As was explained, you can import the full resolution files, or if you're not interested in the improved sound quality of these files for your casual iPod listening, you had the option of converting the large full-resolution files to 128kbps AAC's on import, thereby allowing much more music to be stored on the i-device, but at the lower sound quality of 128kbps lossy-compresssion. The new Apple OS now gives you the option of converting to 128, 192 or 256kbps on import. If you're using very good earbuds or phones, the converters and headphone amp in the iPhone 3GS and above is just good enough that you can discern a slight improvement in sound of a 44.1k/16 bit ALAC and the 256kbps version, but it is not drastic. Since you normally don't use your iPod for critical listening, and are often listening in noisier environments not suited to critical listening anyway, many would rather have more song capacity than the slight improvement in sound. But you can more easily hear the difference in quality between 256kbps and 128kbps. It's audible even on standard-issue Apple earbuds, and on high-quality buds and phones it's very evident. So this new option is perfect for better song capacity on the i-device with only a negligible loss in quality.
To change your hi-res files on import, connect the iPhone/iPad to the computer and launch iTunes. Hi-light your device in the devices list and the Summary page for your device comes up. Scroll down and under Options you'll see an option for "Convert higher bit-rate songs to: " and a pull-down menu with the different bit-rates. Choose the bit-rate you want, then click the Apply button on the right.
Rumor has it that the headphone amp and converters in the iPhone 5 will be a further improvement, possibly the same as are now in the iPad.

Tracking and Mixing in HD - Important News

As I have mentioned in previous posts, there are more and more positive signs that we are moving toward a new push for high-definition consumer audio - meaning 88.2k or 96k/24bit. It's unlikely that any of the bigs will get behind a new hardware format, as they are still stinging from the failures of SACD and DVD-Audio, but it looks like HD audio downloads and streaming are coming on in a big way.
Mastering engineers have recommended 88.2k as being the ideal sampling rate for production if CD's and mp3/AAC's are to be produced, since the sample rate convertor has to do a simple 'linear decimation' to get from 88.2k to 44.1k after mastering (just divide by two) resulting in a slightly better sounding conversion, and in the HD world no one has reliably been able to discern a sound quality difference between 88.2k and 96k. However, it seems there is a major effort happening to make 96k/24 bit the de facto standard for HD consumer audio, so it's advisable that you should be tracking and mixing at that rate. This is not a huge problem, as on an audiophile system you may be able to hear the slight quality improvement between a 44.1k file downsampled from 88.2k vs. 96k. But regardless, the file downsampled from 96k will still sound much better than if the track was produced at 44.1k from the start.
This all makes a difference in the parts you should have your mastering room cut. In the past, of the clients who delivered their tracks to mastering in HD, only a handful were interested in getting the HD mastered versions back - they were only interested in the 44.1k/16 bit master for CD and digital distribution. Now, you should want finished production masters in both HD and CD quality. That way, if the consumer industry does move toward HD audio you are ahead of the curve and have HD tracks ready to offer. Even if that never happens, and we stay in the doldrums of mp3, for your own archive it is best to have your music in as high definition as absolutely possible. The only downside is having to pay for the mastering room to create a second master in HD. This should not be a big expense because if it was done properly the tracks were mastered in HD anyway - the HD versions just have to be exported and written to a data CD-R or DVD-R. Many mastering rooms keep their projects backed up, so if you did deliver HD mixes, chances are good that your mastering room can still create an HD production master for you.

Part 3 - Making Your Computer an Audiophile-Quality Music Player

Let's get outside the computer and talk about things you can add that will greatly improve the quality of your sound. Everything we've done in Part 1 and Part 2 have cost nothing, now let's see about spending a few kerns. If you want to build a real audiophile system, it's best done in reasonably affordable steps, unless you have an unlimited budget. It's a huge trend now to build a desktop audiophile system around the computer. This allows you to go with smaller 'nearfield' speakers, since they will only be 3 or 4 feet away, and consequently you won't need as much power. There are many ultra high-end manufactures who build outstanding equipment just for this purpose. You can also integrate your computer-generated music into your larger home theater/entertainment system, we'll talk about that as well. The following steps are a great way to start building a hi-end system. 

NUMBER ONE

Get a good pair of headphones. That's the cheapest way to start, and you'll love what you hear just by following the steps in parts 1 and 2, getting a good pair of phones, and plugging right into the computer's headphone jack. The headphones and speakers are the final parts that dictate the quality of what you're hearing, so until you improve them you won't really be able to hear all of the improvement we've already made in parts 1 and 2 as well as other changes I'll suggest. Nothing else that you do to your system will make as much difference as adding a good pair of phones or speakers. 

You don't have to break the bank to get into 'reference quality' headphones, but you do have to approach the $100.00 range. Recommending speakers or headphones is tricky business, because they all sound different, and a person's taste in sound is very subjective. So where you might like a bright, upfront, immediate sound, the next person may like a richer, warmer, more ambient sound, and you are both right - it's personal preference. So here's the point; headphones and speakers are not things you should buy online. You need to go and listen to them for yourself. It doesn't matter what pro or user reviews say about them, it's all about whether YOUR ears like the sound, and in the case of headphones, if they are comfortable on YOUR head and ears. However, you can classify these products in terms of accuracy and detail, and that is what this 'audiophile quality' we've been striving for is all about. So it's important to realize that if you're just getting into hi-end 'audiophile' audio, your ears will improve as you listen to good sound, so don't dumb yourself down. Choose among models that are recommended by audiophile websites as being accurate. Just because a phone is expensive doesn't make it reference quality. And a lot of trendy headphones - like the Dr. Dre 'Beats' stuff - will wow you in the showroom by having thunderous bass, but it is way exaggerated and completely overshadows the important upper mids and highs. Developing an audiophile's ear and taste doesn't mean giving up bass, it means learning that tight, punchy bass is what you've been wanting, not enormous clouds of undefined bass. In headphones, this reference quality starts at around $80.00 going up to $1500.00 and beyond. When you go to stores to compare headphones and speakers, take some of your favorite music with you so you're listening to the same thing in each test. 

I'll make one headphone recommendation, and you can take it from there. The Grado SR-80i at $99.00 is probably the best value in true audiophile headphones. The Grado RS-1's that I use in mastering cost $700.00, and you would probably be shocked at how similar these $99.00 phones sound to them. When you get a quality pair of phones like the SR-80i, don't be hasty. If you pull them out of the box and immediately listen to them, you'll likely hate them as the sound will be rather harsh and brittle. Most quality phones (and speakers) require as much as a 100 hour 'burn in' or 'run in' at medium-loud volumes before they develop their smoothness and richness, so put them under a pillow in the spare bedroom with an iPod and let them play for four or five days before you use them. 

Aside from that, the long standing major brands AKG, Sennheiser, Sony, Beyer and Audio Technica as well as Grado offer outstanding choices. Go listen for yourself. 

NUMBER TWO

There are two components in your computer that are reducing your sound quality. First is the DAC (Digital-to-Analog Convertor). The DAC converts the digital audio to analog so your analog ears can hear it. (We are still carbon-based analog life forms, though they're working on that.) The quality of DAC built into even the most expensive computer may be decent compared to DAC's of 10 years ago, but they fall way short of the quality that is available today. A good DAC will deliver, among other things tighter, punchier bass, snappier, more defined mids, a clearer, more detailed top end, and a larger soundstage, which is the apparent width, height and depth of a recording - it sounds more 3-dimensional. 

The second thing that is degrading your sound quality, if you're listening to headphones, is the headphone amp built into the computer. Like the DAC, Apple or whoever is not going to increase the price of a computer by integrating a super high-grade headphone amp; it is just good enough to get by. Adding a high quality pair of headphones will dramatically improve your sound, but you won't really hear what your phones sound like until you match their quality with a DAC and a headphone amp. 

Another factor concerning the headphone amp - some hi-end headphones are extremely power hungry, and standard headphone amps built into computers and iPods can't even begin to drive them properly. The recommendations I'll make will all have plenty of power to drive these monsters. (The Grado SR-80i's are not power-hungry, btw.) Also most of the combo's I'll mention have standard 1/4" stereo jacks to match hi-end headphones. 

The good news is, there are some superb DAC/headphone amp combinations starting around $100.00. And as a bonus, most of them also have standard analog outputs so you can connect them to your preamp or receiver, or to powered speakers. The DAC's I'll be talking about are outboard as opposed to sound cards that go inside the computer. Outboard DAC's will almost always outperform sound cards that cost twice as much or more, as sound cards are heavily engineered for the gaming market, so you are paying more for FX gimmicks and surround capability than you are for sound quality. 

Now we need to learn a little new terminology. An outboard DAC will have one or more of the three standard consumer digital inputs:

USB - standard USB.

S/PDIF - also referred to as 'coaxial', which uses one standard RCA connector (same as the connectors on your hi-fi) for stereo input, one for stereo output. (Although it looks like a standard audio cable, you should get a specific digital cable for connecting to S/PDIF.)

TOSLINK which is an optical connector that uses fiber-optic cables. A single TOSLINK connector can be both an input and an output. (Like S/PDIF digital cables, TOSLINK optical cables are available at any electronics store including Radio Shack.)

And one final helpful term, if a piece of gear has both an input and an output, this is referred to as an 'I/O'. 

The most convenient way to connect an outboard DAC to your computer is via USB, and most computers will immediately recognize the DAC without having to install any additional drivers. The DAC will show up under System Preferences/Sound as a playback option (Mac). On newer Macs, the headphone jack is also a TOSLINK I/O. If you connect your Mac to your DAC using a TOSLINK cable, the Mac knows it's connected to an optical digital input and not headphones - you don't have to set anything except the sample rate in Audio Midi Setup. (The Mac's volume control does not affect the TOSLINK digital output.)

As you look at my recommendations, remember that for best sound the DAC should support the sample rate of the audio file you want to play. And you have to set the sample rate in Audio Midi Setup (Mac) to match the sample rate of the audio you're playing, just as we learned in Part 2. Also, a 24 bit DAC will play 16 bit audio with no problem, but a 16 bit DAC will cause some distortion if playing 24 bit audio. Newer DAC's are almost always 24 bit.

Most people will be absolutely thrilled with 44.1k/16 bit CD-quality audio once they hear it properly. For them, life is easy, as any DAC will play 44.1k or 48k/16 bit. For those who want to explore the hi-resolution world of 88.2k/24 bit and above, things are a little tricky. Due to a completely half-assed engineering kluge in the USB standard, USB does not support 88.2k digital audio. Granted most hi-resolution downloads on sites such as hdtracks are at 96k, but there are many outstanding releases that are in 88.2k (the amazing ABKCO remasterings of the Rolling Stones catalog 1963 - 1970 for instance) However if the same DAC also has S/PDIF or TOSLINK inputs, they will almost always accept 88.2k. So if you don't want the restriction, and if you have a Mac or PC with a digital output, you probably want to get a DAC with both USB as well as TOSLINK or S/PDIF inputs. You actually don't need the USB connection in that case, so if you find a DAC with TOSLINK or S/PDIF inputs only, you're good as long as your computer has a matching output. Rumor is the USB spec will change fairly soon to accommodate 88.2k, and the better manufacturers will likely offer a retro-fit for your DAC, and in some cases it may be a firmware update that you can download and install yourself. 

This is an interesting time to look at DACs. There are some amazing values in all types of high end gear from China. Where Chinese manufacturers used to be just (very good) imitators of classic audiophile designs, they have now established their own technologies and theories of sound and are creating some amazing audio gear at very affordable prices. 

Here are some suggestions for great sounding DAC's and DAC/headphone amp combos:

Filo E7 $90.00 DAC/headphone amp only, no outputs for connecting to other gear. 44.1k and 48k 16 bit only. USB input only.

Filo E10 $80.00 DAC/headphone amp. 1/8" stereo analog line out for connecting to external gear. USB input only, 44.1k, 48k and 96k/24 bit. Amazing sound quality at this price. Cheaper than the E7 but seem to be hard to find.

NuForce Icon uDAC-2 $129.00 A fantastic value, great sounding DAC and headphone amp. USB input only. Supports 44.1k, 48k and 96k 24 bit. Analog outputs on standard RCA connectors for connection to external gear. 

Topping D2 DAC/Headphone amp $160.00 USB, S/PDIF, TOSLINK and analog inputs, RCA analog outputs for connection to external gear. 44.1k, 48k and 96k/24 bit via USB input, also 88.2k via TOSLINK and S/PDIF inputs. Analog input lets you connect analog gear for headphone playback. Chinese DAC/headphone amp that outperforms other high end combos up to the $700.00 range. 

Fubar IV $269.00 DAC/headphone amp, USB input 44.1 & 48k/16 bit, TOSLINK & S/PDIF inputs 44.1, 48, 88.2 & 96k/24 bit. RCA analog out for connecting to other gear. 

Fubar IV Plus $379.00 As above but upsamples to 192k for even better sound quality from headphone jack or line output.

CEntrance DACport $349.00 USB only, but supports 88.2k as well as 44.1k, 48k and 96k, 24 bit. Headphone only, no line outputs. This may seem extreme for 'just a headphone amp', but even the craziest audiophiles agree this is a real world class hi-definition DAC and amp. Unbelievable since it's so small, and the fact that it is totally powered by the USB port yet sacrifices nothing in terms of sound quality, as in others that give you USB-port powering.

NuForce Icon HDP $449.00 This is an excellent value because it is a preamp as well as a DAC/headphone amp, meaning the volume pot controls the RCA analog outputs when no headphones are plugged in, so it can be connected directly to a power amp or powered speakers as a preamp, and it has standard RCA analog inputs so you can connect another piece of analog gear (a phono preamp would be tasty, for instance). The USB input supports 44.1k, 48k and 96k/24 bit, and TOSLINK and S/PDIF inputs also support 88.2k, 176.4k and 192k/24 bit. 

Audio-GD NFB-2 $450 DAC only, no headphone amp. Another Chinese piece that rivals DACs beyond the $1500.00 range. USB input accepts 44.1k, 48k and 96k/24 bit, S/PDIF and TOSLINK also accept 88.2k, 176.4k and 192k. 

CONVERT YOUR VINYL TO DIGITAL AND MAINTAIN THE ANALOG SOUND

If you have a large record collection, you might consider an ADAC (analog-to-digital/digital-to-analog converter), meaning it can be used to record as well as play digital audio. The ADAC will have standard RCA analog inputs where you would connect a phono preamp or the recording outputs of a receiver or preamp. Like the DAC, these will connect to your computer via USB, TOSLINK or S/PDIF. You will also find many affordable applications for recording your records that will help you remove clicks, scratches, surface noise and rumble. 

The Furutech GT-40B at $525.00 is really an excellent value. It's an ADAC with both analog line inputs AND an excellent phono preamp (MM and MC), as well as a super hi-end headphone amp. This USB-only ADAC will allow you to record (and play) at 44.1k, 48k or 96k/24 bit. Recording your albums at 96k/24 bit, in combination with the superb quality of the convertor, will maintain all of the richness of your vinyl's analog sound.

You will see many 'USB turntables' on the market that have an ADC (analog-to-digital converter) built in, so all you have to do is plug them into your computer and go. They often even include recording software. However, you are getting neither a good turntable nor a good phono preamp nor ADC in these things. If you want to maintain the quality of your albums' analog sound, you need a good turntable, good cartridge, good phono preamp and good ADC.

When you record your albums, you will be creating standard AIF or WAV files, which we learned about in Part 1. Your best albums should be recorded at 88.2k or 96k/24 bit. Others at 44.1k/24 bit will still sound excellent while taking up half the space. As we talked about there, these files will be huge compared to mp3's, AAC's or even CD-quality 44.1k/16 bit files you've ripped from your CD's. So after recording and de-noising an album, you would use iTunes or the XLD utility to convert the AIF or WAV files to one of the lossless-compression formats Apple Lossless or FLAC, after which they will be almost half their original size. And of course you could create mp3's or AAC's if you prefer those for your iPod. 

That is way too much for Part 3. In Part 4 we'll continue this thread and talk just a little about amps and speakers, and options for integrating your computer audio into your larger entertainment center, including wireless options. We'll also talk about software and 'asynchronous' DAC options that will take your system over the top of true audiophile sound.