If you have an extensive vinyl collection, you may be considering the idea of digitally recording your vinyl records. Most of the research I’ve done in this area seems to send me down one of two paths: 1.) El cheap-o USB turntables, or 2.) Exorbitantly expensive audiophile gear. It’s tough to come up with something that’s very good, but won’t break the bank. In this guide, I’ll make some suggestions and offer some ideas to consider so that you can feel confident that you’ve done your precious vinyl and, perhaps more importantly, your precious time justice without wasting a ton of money on nuances that only a very few would ever notice.
It’s important to grasp a few basic concepts before you proceed. Keep in mind, on any of these concepts there is an immense amount of knowledge to be had. I’m going to give you the “Readers Digest version.” If you’d like to study more, that’s wonderful. But keep in mind, it’s easy to go down the rabbit hole. There are lots of blogs and discussion forums where people discuss the minutiae ad nauseam and it’s really easy to get bogged down. Try to appreciate the broad strokes and keep them in mind as you move forward.
Dynamic range is the term used to refer to the difference between the loudest sound and the softest sound that any given recording medium can record. It’s measured in decibels. Vinyl records typically top out at about 70 decibels of dynamic range. A digital recording that you make with your computer will typically have a dynamic range that is significantly greater, but it does depend on the settings you use. More on that later. Just understand that in order to do your vinyl records justice, you’ll want to record them using settings that will most certainly capture every bit of the dynamic range that exists on your vinyl record.
Digital recordings aren’t continuous like analog recordings. In the digital world, you take thousands of snapshots of what things sound like each second. When you string those snapshots together you can record the sound waves over time. The rate at which yo do that is called the sample rate. A compact disc has a sample rate of 44.1 KHz. That means that it’s taking a snapshot of the sound wave 44,100 times each second. The higher the sample rate, the wider the range of frequencies can be recorded. The 44.1 Khz sample rate is the lowest rate that will cover all of the frequencies that humans can hear.
Bit Depth defines how much dynamic range we can record. A compact disc uses 16 bits. That’s about 96 decibels of dynamic range. Increasing your bit depth to 24 bits allows you to record about 144 decibels of dynamic range. As you can see, even 16 bits is ample bit depth to represent the full dynamic range of a vinyl record, and then some. But there are some reasons that you may want to record at 24 bits which we’ll discuss later.
To learn more about sample rate and bit depth, check out this fantastic article.
Cartridge / Stylus
The cartridge you choose is arguably one of the most important choices you’ll make when choosing what gear to buy. After all, it’s the component that’s actually coming in contact with the record. You can spend as little as $40 or as much as $10,000. Which cartridge is right for you? This is a tough one. How can you know how good a cartridge sounds? Unless you have a listening room and the ability to hear them compared to each other while looking at test equipment, you’ll probably do what most people do: you’ll read reviews and buy the best thing you can afford. This is quite a rabbit hole. Spherical or elliptical? Moving magnet or moving coil? Every audiophile internet forum is full of passionate, dogmatic opinion. I won’t subject you to my opinion other than to tell you that I use a Nagaoka MP-150. It’s an elliptical stylus on a moving magnet cartridge and it runs about $330 on Amazon. I think it sounds fantastic. It’s not cheap, but it’s also not a gajillion dollars either.
In my opinion, we’re living in a time where even a low-end turntable isn’t awful, especially if you’re not trying to do anything fancy with it like scratching, mixing, etc. But there are a few things to consider. It may be temping to go with a “USB turntable.” They’re affordable, usually around $130-$150. They have a built-in preamp, cartridge, and analog-to-digital converter. You plug it right into your computer and off you go. Honestly, if that’s what you want to do, it won’t sound horrible and I won’t try to talk you out of it. The purpose of this article is to try to make it better but still keep it reasonable.
I use a Technics SL-1200 mk5 which I bought because I was a DJ for many years. If I weren’t DJing and wanted a nice turntable that’s solid quality and easy to set up and configure, I’d consider the Pioneer PLX-1000 ($700) or the Reloop RP-7000 MK2 ($500.) You want something that has a standard connector for headshells, the ability to install any cartridge you wish, the ability to balance the tonearm and an anti-skate dial.
Friendly reminder… Don’t forget to set your pitch control to 0%. You can always manipulate it however you wish in post, but you want your recording to represent that vinyl disc as faithfully as possible.
The signal that’s generated by a phono cartridge is minuscule. That’s where a phono preamp comes in. It takes that tiny little signal and boosts it so that it can be heard by downstream equipment. It also performs another crucial role. Starting in about 1954, virtually every record created uses something called RIAA equalization. According to Wikipedia, “The purposes of the equalization are to permit greater recording times (by decreasing the mean width of each groove), to improve sound quality, and to reduce the groove damage that would otherwise arise during playback.” In order for your record to sound normal, the preamp reverses the effect of this equalization. Just like any audio gear, you can spend a ridiculous amount of money on a preamp. I mean hey… why not by the Burmester 100 for $27,000? I went with Schiit Audio’s Mani for $129 and I think it sounds great. You plug your turntable’s cables into the input, connect the ground wire to its terminal, and out comes line level audio. Whatever preamp you choose, make sure it’s configured properly to work with your cartridge. Do some test recordings. Listen carefully on different settings and look at the wave forms in your audio editing software to see if there’s obvious clipping.
Analog to Digital Conversion
Finally, you’ll need something to turn the line-level analog signal into a digital signal that can be captured by your computer. This is called an analog to digital converter or “ADC.” You could use your computer’s on-board audio interface for this task. If you do, in most cases, you’ll be limited to 16 bit and either 44.1 KHz or 48 Khz sample rate. For reasons I will discuss later, I decided to purchase an ADC that was capable of better sample rates and bit depths along with circuitry built to a higher standard than what would be baked into a garden variety computer sound card. I chose the Focusrite Scarlett 4i4 for $230.
Before you play those records, you’ll want to make sure you give them a good cleaning first. I bought the Spin-Clean Record Washer MKII Deluxe Kit. It’s criminally overpriced in my opinion, but otherwise I have no complaints. Watch some YouTube clips on how to use it an you’ll be good to go.
If you prefer a different method, there are many, many to choose from. Some wash them in the sink with Dawn, some use their favorite combination of brushes and devices. Some use a vacuum cleaner attachment. There’s even a method that uses wood glue!
For your stylus, I recommend Audio Technica AT607 Stylus Cleaning Formula. It has a built-in brush on its applicator.
So you’ve got all your gear. Now what? You don’t really want to have to do this again, so carefully consider each step and each decision you make.
If you’ve ever recorded something to an analog medium like a cassette tape for example, you probably know that to get the best quality recording, you must set the gain to get the maximum amount to signal onto the tape without overdoing it lest you get distortion. That’s because with analog audio, there’s always noise in the background and you want as much signal as possible so that you don’t notice the noise. This is referred to as a good signal to noise ratio or SNR. The beautiful thing about recording digitally, is that there is effectively no noise. So you don’t have to worry so much about getting the signal’s gain to that sweet spot. As long as you don’t create a condition where the gain is so high that the signal clips, then you’ve got yourself a good recording. Afterwards, you can adjust the loudness without affecting quality at all.
Choose a Sample Rate and Bit Depth
I’ve chosen 96 kHz and 24 bit for my vinyl recording. Why 96 kHz when 44.1 kHz will record the entire frequency range that humans can hear? The only reason is that if I decide to apply a filter in post, having the extra bits can make some filters do a better job. Plus, in today’s day and age, disk space is cheap, so I decided to go with the higher sample rate. If you decide to stick with 44.1 kHz or 48 kHz, you can feel just fine about it knowing that your recording will sound great. If you’d like to drill down on this more, sonicscoop.com’s Justin Colletti has a nice explainer. Why 24 bits (144 dB) when a vinyl record’s dynamic range is only 70 dB? What if the output of my preamp is low enough to make the quietest moments drop below the -96 dB limit in a 44.1 kHz recording? Not likely. But again, disk space is cheap and with 24 bits, I know I’ll have more than enough dynamic range to do the job. Can you feel good about recording at 16 bits? Yes, I think so.
There are so many options, choosing one can be daunting. You can’t go wrong with Audacity. It’s free, it runs on all platforms, and the audio quality will be every bit as good as the most expensive products out there.
I use Amadeus Pro on the Mac. I prefer its interface and since I’ve been using it for many years, I’m accustomed to it and can work more efficiently in it. It also has an excellent interface for manually repairing clicks which I’ll explain later. One of its biggest flaws is that when converting bit depth, it doesn’t perform dithering. We’ll get into that later as well.
So you’ve got your gear all set up and configured and you’ve got a squeaky clean record and stylus. It’s time to start recording. Make sure your turntable is on a stable surface that is isolated from vibrations as much as possible. While recording, it’s important to not create vibrations that might make their way into the recording. If you have to walk around, tiptoe as softly as you can. Don’t work on the surface your turntable is sitting on. Even faint bumping and tapping can make their way into your recording. Keep speaker volume to a minimum or use headphones; vibrations from nearby speakers can affect the recording or even cause feedback if loud enough.
When you’re done recording one side of your disc, hit pause on your computer and record the other side of the disc into the same file so that the entire record lives inside of a single audio file. Why? Keep reading. When you’re done, save your file in an uncompressed format. AIFF or WAV are appropriate. I choose WAV because it’s more common and probably has more support on different platforms and in different software. Eventually, you’ll want to convert them to MP3’s, AAC’s, or FLAC files. But hang on to those originals; you may decide you want to go back to them someday.
After the recording is finished, you’ll want to normalize the file. When you normalize, you’re taking the loudest part of the recording and boosting it so that it hits at or near the maximum possible value. This doesn’t affect quality; it just raises everything so that everything is louder. Some considerations about normalizing:
- Your record is designed to be an entire listening experience. One side of your record might be a little louder or softer than the other. If you were to normalize the two sides separately, you may wind up with some tracks being louder than they’re supposed to be relative to the others. That’s why it’s a good idea to put the entire record into a single file and normalize the entire thing.
- Your recording may have clicks and pops from damaged portions of the record. One of these clicks might be the loudest part of your recording and you really don’t want a click to define the loudest moment. You want the loudest moment in the music to do that. So before you normalize, use your software to find the loudest moment. In Amadeus Pro, you do a command-A to select all, go to the “Analyze” menu and choose “Find Maximum.” Amadeus will move your cursor to the loudest moment it can find. Listen to that moment. If it’s a click, fix it and then search for the next loudest moment. Keep doing this until the loudest sound is not a click. Then, it’s safe to normalize.
In Audacity, there doesn’t seem to be a way to move the play head to the loudest moment. There’s a plugin that will scan across the file for the loudest value and report what it is, but it doesn’t take your cursor there so you can hear it. If you know of a way to do this, please leave a comment and I’ll update this tutorial.
When you really start to listen to your beautiful new recording, you’re going to notice some things you may not like. No matter how much you clean your records, you’ll still hear some clicks, pops, and crackles. You’ll also hear surface noise. How much you decide to mess around with the file once you’ve recorded it depends on what you perceive to be an imperfection, how much it bothers you, and how much of your time you perceive to be justifiable in repairing it. After all, you’re going to be the one listening to it
There are many tools available to mitigate these phenomena, but none is perfect. And I have found that in order to fix things properly, there’s a pretty significant time investment.
You can fix clicks manually or by using an automated process. “Why would I want to manually remove clicks when I can get software to do it for me?” you ask… I have found that even the state-of-the-art automated click repair algorithms have major problems with false positives. Many percussion elements, particularly electronic drum machines, will look like a click to an automated click-removal algorithm. It would seem possible to write an algorithm that would be able to detect a rhythmic pattern of clicks and leave them untouched while repairing only those clicks which occurred with a sufficient degree of randomness. iZotope’s $400 RX software has a de-clicking function that claims to have “a protective algorithm for preserving periodic audio elements.” It sounds like it’d be great, but it doesn’t work.
Here’s a passage from Hashim’s “Al-Naafiysh.”
iZotope’s RX has the ability to preserve just the clicks. Listen to what it finds on its default setting.
Notice that the attack on the kick drum and rimshot triggers RX’s algorithm. Here’s what it sounds like after RX has removed clicks.
This is a show stopper. Clearly, there’s a non-random pattern to these clicks. Despite iZotope’s claims, its “protective algorithm” is, for all intents and purposes, useless.
Audacity has a click removal function as well, but it doesn’t have the ability to play the sounds that it removes. Without the ability to effectively scrutinize the algorithm, it’s not a useful tool in my opinion.
Sadly, automatic click removal can have a significant detrimental effect on the music. And, in my experience, it doesn’t just happen on electronic sounds either. I have had some success using it to remove clicks where no percussive elements are present and during the fade at the end of the song where clicks are very loud relative to the music.
Here’s a passage that contains no percussive sounds that might generate a false positive, the intro to “Let’s Jam (Dub Version)” by Newcleus. Notice the severe clicks due to record damage.
Here’s the same passage after applying iZotope’s RX de-click and de-crackle functions. (Default settings.) This is where RX shines. The results are fantastic.
Here’s Audacity’s attempt. (Default settings.) It seems that only the most severe clicks are affected and it seems to only reduce their severity rather than eliminate them.
If you’re like me, you may find that, in many situations, performing manual click repair is the only effective method. The real cost here is time. How much time do you want to spend listening to your track, finding every click and repairing it? I suppose every person strikes the balance between how much they care and how much time they’re willing to spend differently. I would recommend keeping the original unedited file so that if you want to do it again in the future using a different method, you can. I would also keep the edited uncompressed file so that, after listening to your music a few times, you can go back and fix parts that you find particularly bothersome.
Hairersoft’s Amadeus Pro ($60) is my preferred tool for manual click repair. Amadeus Pro employs hardware accelerated video so that zooming in, zooming out, and playing the waveform in real time is very responsive. (Much better than iZotope’s RX.) It has a feature called “Repair Center” which is very useful for manual click repair. It displays an area near the cursor and super-amplifies the waveform so that tiny clicks are easy to see.
Once you have selected a click, you can press “R” to repair both channels, “T” to repair just the top, or “B” to repair just the bottom. When it repairs a click, it does a fantastic job in my opinion. When listening to it after the repair, I have never heard an artifact that bothered me; it just removes that click. “N” takes you to the next click that it finds, although I have found that its ability to find the next click leaves something to be desired. It often identifies sections that aren’t actually clicks, or skips over sections that contain obvious clicks. If you click the right side of the window, your view will advance forward and vice versa. Performing this task effectively will take practice.
Once you have your recording all cleaned up, it’s time to encode it.
If you have recorded in a higher bit rate and/or sample rate, you’ll want to convert your file to 44.1 kHz or 48 kHz. It’s important to use software that does this conversion with some sophistication. You’ll also want to convert the bit depth to 16 bits. To do this properly, you’ll want to use a program that employs dithering. Audacity employs a dithering algorithm. Sadly, Amadeus Pro does not, but the developer is planning to add this functionality in the next major release. iZotope’s RX employs their own “MBIT+” dithering algorithm. Most high end DAWs have some sort of dithering functionality baked in.
Once you have a 16 bit, 44.1 or 48 kHz file, you’ll want to choose a format that’s useful for the various devices that might play it. Uncompressed audio files are definitely what you want for your archive, but they’ll take up too much room on your phone. Plus, WAV files don’t have the best support for meta data. There’s already so much written about how to encode audio files, I won’t go too deeply into it; I’ll just summarize the most common choices.
- Lossless formats – This includes FLAC and ALAC. If you’re looking to preserve every bit of the quality of the original, you’ll want to use a lossless format. If you’re using Apple products, you’ll probably want to use ALAC since it’s supported in iTunes and in all of Apple’s mobile devices. FLAC has wide support elsewhere. There are numerous utilities that will transcode between the two if you need to and since they’re both lossless, doing so will not affect sound quality. You may also run across Monkey’s Audio “APE” format. Its sound quality is every bit as good, but good luck finding a mobile device that will play it.
- Lossy formats – This includes MP3 and AAC. MP3 is by far the most ubiquitous. If you want to encode your file using MP3, using a bit rate any less than 320 kb/s will negatively affect high frequencies. MP3 can be played on just about any platform. AAC is the codec used by Apple for the iTunes store and its Apple Music streaming service. This codec is generally superior to MP3 and yields better sound quality at similar bit rates. It’s well supported on most modern platforms, but you may run into the occasional device that won’t play it. If you know that you’re going to remain in an Apple ecosystem, AAC is a good choice. You may also run across OGG files which are just fine, but are far less supported.
If you’re using a Mac, XLD is the encoder of choice. It’s freeware. It has many options and you can be sure that it’s doing what it’s doing at the highest possible quality. If you’re using Windows, you can download an MP3 encoder for Audacity or you can pick from a myriad of other options.
Share Your Ideas
If you’ve found something that works for you, (or doesn’t work for you,) and you’d like to let others know, please comment on this article. There’s always a better way. Got questions, post those too. I’ll do my best to answer.