User:Mjb/Ripping vinyl

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Here are some random notes about my vinyl ripping process.

Physical connections

For vinyl, unless you have a modern turntable (phonograph) with USB output, you need to connect the analog outputs of the turntable to a preamp. The preamp boosts and properly EQs the very weak input signal coming from the turntable so that it's at the standard "line level" expected by other devices.

Examples of preamps:

If the preamp doesn't have USB output, then you need to connect the preamp's line-level outputs (e.g. "tape out", "rec out", or "monitor") to the audio "line in" inputs on your computer.

If you are ripping tape instead of vinyl, you don't need a preamp; tape decks have line out connectors which can go directly to your line in.

All of these connections use standard "RCA"-type patch cables, the same kind used by TVs, VCRs, video game consoles, and consumer audio gear since the 1950s. You do not need to buy expensive cables, but it does help if they are well-made and have good shielding, otherwise you risk them picking up EM interference. If you keep hearing the sound of a local radio station or noises from cell phones, you probably need better cables. Otherwise, they are fine. In my experience, 4 out of 5 random cheap cables are perfectly OK.

DJ mixer connections

I am using a DJ mixer as my preamp. In addition to the crossfader and input level controls, it has 3 tone controls and a mono/stereo switch, plus an internal amplifier with a volume knob. The mixer has two line outs. The "rec" output bypasses the mixer's internal amp, while the "amp" output carries the processed audio. So I have the rec out going to my computer's soundcard, and the amp out going to my external sound system.

When ripping, I don't use the external sound system at all; I don't want any loud noises or bass vibrations feeding back into the needle.

I also try to ensure the mixer is only outputting sound from the one phono source. The crossfader is all the way to one side, the slider for the phono input level is set as high as it can go without distorting (depends on the loudness of the record, but usually around 70%-80% of maximum), and all the input levels of the other channels are at zero. More on this in the testing section below.

Recording software

To do the recording, you need audio editing & recording software, a.k.a. a wave editor. I use Adobe Audition, which has a lot of good features but is not free. A lot of people use a free alternative called Audacity. If you were to use Audacity, it requires some extra configuration to disable the default addition of "dither" (noise like tape hiss) to all the files you save with it, so be careful with that.

Configure the software to get its input from the appropriate device, and to record in stereo.

Testing my input levels

In Windows, in the Sound control panel, Recording tab, (Realtek High Definition Audio) Line In Properties, Levels tab, the Line In slider can set the input gain from 0 to 100. What do these numbers mean?

Here is how I figured it out:

My soundcard only supports 16-bit recording. If I have Audition record at 24-bit, the lowest 8 bits of each sample are random numbers, just noise (I guess from the Windows mixer), so I record at 16-bit.

With no incoming signal (DJ mixer powered off) and the Line In slider at 0, the background noise is extremely low. Sample values range only from -1 to +1, with the average RMS of my line noise coming out to about -99.3 dB (as measured by Audition's Amplitude Statistics window with "0 dB = FS Sine Wave" and "Account for DC" checked, plus "Window Width: 50 ms").

As the Line In slider goes up from zero, the level jumps, but the higher it goes, the smaller the jump. Here are the average RMS levels for various slider settings, and the sample value min/max. The last two columns are for when the mixer and turntable are powered on, but nothing is playing. These values will vary slightly depending on line noise conditions.

  0 -99.3   -1/+1    -94.4   -2/+2
  1 -94.2   -2/+2    -90.5   -3/+3
  2 -91.4   -3/+3    -88.5   -4/+4
  3 -89.9   -4/+4    -87.5   -5/+5
  4 -87.5   -5/+5    -83.1   -8/+8
  5 -85.8   -7/+7    -85.2   -8/+9
  6 -84.1   -7/+9    -81.7  -10/+10
  7 -83.6   -8/+10   -79.9  -11/+12
 50 -68.2  -50/+64   -64.4  -69/+75
100 -62.4 -102/+127  -57.3 -139/+165

I suspect the positive skew is just due to my electrical line noise being a little stronger in one channel.

So, for this particular soundcard, the numbers are how much you want to amplify the input to the ADC, on a logarithmic scale. That is, zero does not change the input level at all, 1 boosts it about 5 dB, 2 boosts it about 8 dB, 3 boosts it about 10 dB, and so on, with 100 boosting it about 37 dB. So unless you are dealing with abnormally quiet signals, you want to keep this control set very low. However, I found that it's not adding any more noise at higher levels (e.g. when I record at 0 and amplify by 37 dB, the noise is exactly the same as recording at 100).

As I mentioned, I have to keep the DJ mixer's phono input level at about 7 out of 10, or the bass hits on a fairly loud drum & bass 12" will start to clip, regardless of the input gain level. This suggests that the mixer's sliders can amplify the input from the turntable, not just attenuate it.

With the mixer's slider at 7, and the mixer's volume knob at the halfway point, these bass hits peak at 0 dB on the mixer's VU meters. VU meters are a crude analog metering system showing averages, not instantaneous peaks, and where zero is roughly -18 dBFS. Indeed, with the soundcard's input gain at 0, Audition shows the loudest instantaneous peaks are about -14 dBFS, although most of them are down around -18 to -22 (I'll just say -20).

Conclusion: If I set my mixer's master volume at the midpoint, I can use 0 dB on the mixer's VU meter as an ideal peak for what's going out of the mixer. On the computer, I settled on setting the Line In input gain control at 7, which gives about a 13 dB boost and thus makes the actual peaks be about -7 dBFS in Audition, plus or minus a few dB, depending on the loudness of the record.

The goal is to get the peaks to be no higher than about -3 dBFS, at which point they might be getting compressed. Most soundcards have built-in dynamic range limiters to protect their circuitry from excessive input voltages. You can see them in action if you crank up the volume coming into the soundcard, and it just squashes everything into the 0 to -3 dBFS range and never clips.

Update: If I configure Audition to use ASIO input (after installing ASIO4ALL, which provides ASIO wrappers to WDM APIs to bypass the Windows mixer), the input gain control doesn't do anything. However, I get a signal equivalent to setting the control to 6. Why? I don't know!

Input levels

Some decks, preamps, mixers, and computer soundcards may have input volume (gain) controls which need to be adjusted to ensure the signal is as strong as it can be without overloading and distorting.

These devices also may have built-in dynamic range compressors/limiters which protect from overloading but flatten out the sound. My computer's soundcard, for example, begins flattening the input when the level gets within 3 dB of the maximum, so I try to make sure the loudest parts don't exceed that level.

Know where your OS's audio controls are and what they all do. Disable or set to zero any special audio enhancement features of your hardware; this includes stereo/mono switches and tone (bass/mid/treble) controls.

Sample rate and bit depth

The sample rate and bit depth to capture must be chosen. Higher values use more space.

Set the same values in both the software and your OS's sound hardware controls; otherwise it may refuse to work or it may just silently convert to what you asked for in the software.

The choice of sample rate affects the audio frequency (pitch) range. The upper limit will be slightly under half the sample rate, e.g. if you sample at 96 kHz then you will get pitches up to about 48 kHz. Most records have nothing but noise above 30 kHz, human hearing maxes out at 20 kHz, some amplifiers distort lower frequencies when fed ultrasonic frequencies. So for pretty much anything you ever record, the standard 44.1 kHz or 48 kHz are good sample rates to use. 48 kHz is standard for audio synced with video, and 44.1 kHz is the only rate supported by audio CDs. So basically you just choose 44.1 or 48. People who misunderstand the science behind it tend to think higher sample rates must be some kind of improvement on the audible range, but it really just wastes space capturing ultrasonic frequencies.

Bit depth determines the dynamic range (number of possible volume levels, basically) and also the amount and volume level of "quantization noise" (raspy, buzzy, distorted sound) resulting from the rounding of fractional levels to whole numbers. Modern soundcards can only do 16-bit or 24-bit sampling, and at these levels, quantization noise is not an issue unless you are recording the sound of butterfly wings flapping in an anechoic chamber. An empty record groove or blank tape will have a noise floor at around the 9 to 12 bit level, and all the music will be well within that range, so 16-bit is fine and is standard for CD. Again, people think higher must be better, and 24 would give you more room to use very low input levels or capture the sound of those butterfly wings, but for recording from vinyl or tape it's just wasting space. Lowering the input levels also reduces the signal-to-noise ratio (SNR).

You may also see 32-bit floating-point sampling is an option in your software. This doesn't hurt, but it is not higher quality; the hardware is delivering, at most, 24-bit, and the software is just saving it in a way that removes some limitations which are of no consequence to you; it's like adding a decimal point and zeroes to each number. In fact, even if you tell your software to record at 16 or 24 bit, it actually records at 32 temporarily, then simplifies the numbers when saving. (There is a bug or feature in Adobe Audition, though, where if you tell it to record at 16 or 24, it saves with dither, regardless of the dither setting, whereas if you record at 32 and then save as 16 or 24, it honors the setting. For vinyl ripping this is of no consequence.)

Recording and editing

Once you have set up everything, you just hit your software's Record button, and start playing your record or tape. Stop when you're done, and save your work in progress.

After you have captured some audio in this way, it probably needs some editing:

  • splitting into separate files
  • conversion to mono if the input was mono
  • trim the ends; maybe fade in and fade out
  • adjust the overall volume
  • adjust the left/right balance by adjusting volume or using EQ in one channel
  • apply a small amount of EQ to correct for deficiencies in the recording chain (maybe a slight treble boost)
  • reduce noise (see next section)

To keep the sound true to the original, I avoid the temptation to use "remastering" types of effects like heavy EQ, stereo expansion, or any kind of dynamic range compression.


I've noticed that using a hi-fi mono cartridge & stylus (the one I use for 78s) can make some mono 45s sound better than they do when using a stereo microgroove cartrdige & stylus. It seems to depend on the record. I am guessing the mono cart's needle just rides up a little higher in the groove (its tip is 3x wider than the stereo needle's tip) and thus on these records may be avoiding some of the most damaged parts of the groove walls.

Sometimes a stereo rip of mono material comes out better if you convert to mid/side and just use the mid, or if you use the better-sounding channel of the left and right, or if you aggressively declick the side and then sum to mono. Pick whatever sounds best.

Reel-to-reel notes

When recording from reel-to-reel tape, I found that many of the tapes are recorded with 1 mono track per side, but a stereo player plays both sides at the same time, with the left channel being the front side and the right channel being the back side, playing in reverse.

This saves a lot of time since both sides play at the same time, but it means I have to use my software to reverse the right channel and save it separately from the left, and to make each file be centered mono. When doing this, I may also change the sample rate, depending on the content. I use the software's spectrogram view to visually see what frequencies are in the signal, and I decide whether to resample to 32, 24, 16, or even 8 kHz in some cases. If there's nothing but noise above 9 kHz, for example, I'll resample to 24 kHz so that 12 kHz will be the upper limit. Reducing the sample rate cuts out the extra hiss and makes the audio take up much less disk space. Resampling is safe to do in the software I use, but some software does it poorly and messes up the sound. is a good site to see which apps do a better job.

78 notes

78 RPM records are usually made of brittle shellac instead of vinyl. Almost without exception, they are in mono and have a very wide, deep groove. Unlike vinyl, you can handle their playing surfaces with your clean, dry hands without having to worry about ruining them.

To play a 78 in the authentic way, you need an olden-timey phonograph which probably uses a very heavy tonearm (they were especially heavy before the mid-1930s), and either a steel or tungsten needle, with the sound of the vibrating needle going through a horn for amplification, no electricity involved other than for the platter motor. With this system, both the needle and the record will wear out very quickly.

Cartridge and stylus

You probably actually have a relatively modern turntable which normally uses a magnetic cartridge with a diamond or sapphire needle. A normal, relatively tiny stereo needle and cartridge made for playing "microgroove" vinyl will sound terrible when you play a 78 with it. For proper "coarse groove" playback on a modern turntable, you have to use a special type of magnetic cartridge and needle, like this: (There are some cheaper ones available, but they are not ideally made; they are stereo cartridges which respond to the depth of the groove but are wired to only output the electrical "sum" signal. Ortofon has a good explanation of this.)

A typical stylus for 78s requires a tracking weight much higher than you would use on vinyl. My MONO3-SP seems to do OK at 5 grams. My turntable can only do anti-skate to 3 grams, though, so I probably am riding one groove wall slightly harder than the other, and it may be more prone to skip. So far, so good, though.

My 2.5-mil stylus may be a little noisier than is ideal for some records. At Roger Beardsley suggests that a 3.0-mil or 3.2-mil stylus would track Victor 78s with less noise. I assume this is because there is an optimum height along the groove wall that you want the tip of the needle to ride along, and this height is surely also affected by how worn the walls are at different heights after having been played with various needles over the years. I would not be surprised if, from record to record, it's not quite as consistent as Beardsley implies. Also I noticed that the choice of different styli used in the George Blood digitizations at doesn't have a tremendous impact on the sound!


Many of my 78s are actually fairly clean already. I tried several budget methods of washing and drying some of the records, even vacuum-drying, but the results were not really worth the effort. Crud in the groove tends to be stuck in there really good. The worst of the surface noise is from wrecked groove walls, anyway; it is a scrape and crackle that gets louder when the music gets louder, and cannot be fixed by cleaning.

I had one record which was not pure shellac just quietly break up into a half-dozen pieces when I used hot water on it. The water wasn't even all that hot. Luckily it was not a rare or interesting record, but still... be careful with 78s! And don't let alcohol based cleaning fluids anywhere near them.

Speed and frequency content

Ideally, your turntable also needs to be able to spin at 78 RPM, although it will work just as well* if you use 45 RPM and then rewrite the sample rate—that is, you can play the record at 45 RPM and sample it at 45/78ths of your soundcard's sample rate, and then tell your audio editing software to pretend it's the soundcard's rate. For example, play the record at 45 RPM, have the soundcard set to read at 48000 Hz, tell the software to sample it at 27692 Hz (this means relying on your software to do good on-the-fly conversion from the hardware's rate), then make your editor interpret the sample rate as 48000 Hz. This is not the same thing as sample rate conversion or DSP-based pitch/time adjustments; you are not modifying the audio at all, just the declared rate in the file. Everything is just shifted upward, so what was 45 Hz is now 78 Hz, 4500 is now 7800, and so on, on up to 27692 which is now 48000.

Despite their high rotation speed, most shellac 78s were recorded with no frequencies above 10 kHz until around 1937. In the acoustic recording era (up to about 1926), there was rarely anything below 200 Hz or above 4 to 6 kHz. Spectrograms of my own rips show a variety of frequency ranges in some of the electrical recordings, including what appear to be some "ghost" harmonics up to 16 kHz, and mysterious noise-level jumps correlated with louder, more treble-y parts of the music.

The phono preamp will apply the standard RIAA EQ curve which is designed for the majority of vinyl LPs made after c. 1954 (sources vary) although anything before 1965 could have a slightly different curve expected. 78s used different EQ curves, with very little standardization. They also were not cut at exactly 78.0 RPM. There is a lot of research online about this, but unfortunately for any given record, there is not going to be a lot of agreement about what the precise playback speed and correction EQ really should be, and you may prefer the sound with a more aggressive EQ than what is prescribed.

  • acoustic recordings (–1925) = varies wildly; check online sources
  • electrical recordings (1925–) = absent better info, assume 78.26 RPM; 45/78.26*48000=27600
  • Columbia acoustic = 80.00 RPM; 45/80*48000=27000
  • Victor acoustic (1909–1926) = 76.59 RPM; 45/76.59*48000=28202; music in 160–3300 Hz range
  • Victor/Victrola electric (1926–1931) = 78.26 RPM; 45/78.26*48000=27600; music in 59–10200 Hz range (though may only have false or very weak harmonics above 5 kHz)

On Victor records, the trademark Orthophonic means electrical recording: the original wax-coated disc from which the molds for the shellac 78s were derived was cut using a signal from a Western Electric microphone. These recordings have better frequency response than the horn-based acoustic recordings. The electrical recordings were first made in Spring 1925 and were soft-launched on records identifiable by a triangle after the matrix number, or more commonly by an oval VE stamp in the runout area. Finally in October 1925 the Orthophonic name was publicly introduced by Victor as 1. a special line of Victrolas (record player/cabinet units) designed for optimum playback of the electrical recordings, and 2. a name for the electrical recordings themselves. The labels of the records were redesigned to include VE at the top and to mention Orthophonic Recording on the side. An example of a V.E.-stamped record being repressed with Orthophonic Recording labels:

Some Victor records have an interesting signal in the HF noise. The intensity and frequency varies from record to record, side to side. Sometimes it sounds like aliasing noise, varying in intensity with the music. It usually starts high and then sweeps slowly lower in pitch (maybe after going upward for several seconds). By the end of the record, it is still above the highest frequencies of the music. I have not determined if this is only on electrical recordings or if it is present in acoustic too.

Edison discs

Edison "diamond discs" are 80 rpm and look like 78s, but are about 3 times as thick and aren't made of shellac. They also are vertically cut, i.e. the groove is a perfect spiral and only the groove depth modulates. To record these, use a stereo needle. In an audio editor, convert from stereo to mid-side, and just take the side channel. The mid channel should be nothing but noise. If when listening in stereo it sounds like there is an echo with one channel delayed by one rotation (¾ second), you need a different needle. Supposedly an ideal needle is 3.5 mil, but I got much better results from a 0.7 mil shibata in good condition than from a worn 0.7 mil elliptical; the elliptical gave me very loud echoes as if it was playing two grooves at once!

78 ripping procedure

Here's my procedure so far, roughly:

  • with soundcard configured for 16-bit, 48000 Hz (its maximum), have audio software record at rate from list above, 32-bit float
  • apply inverse RIAA EQ (must be done first)
  • interpret sample rate as 48000 (this fixes the speed)
  • normalize to -3 dB (my 78s record at much lower levels than vinyl, and I don't feel like adjusting the soundcard's gain)
  • trim ends (can be done at any time, really)
  • further EQ according to online guides and taste
  • optional resampling to save space

Thoughts on speed and EQ guides for 78s

The online guides I found for using EQ to correct the sound of acoustic recordings on 78s played with magnetic/moving-coil cartridges seems to be:

  • partly "earballed" many decades ago based on a handful of recordings from each label;
  • partly based on theory (amplitude rising in proportion to frequency);
  • expressed solely in terms of electrical EQ, "bass turnover" and "treble rolloff" being the only two parameters.

However, my analysis & experimentation with files from a 2014 re-creation of an acoustic recording session suggest it's not so simple; a corrective EQ for that session was something more like this: 100 Hz and below shelved @ +15 dB, 375 Hz to 1650 Hz shelved at @ -9 dB, 2500 Hz and up shelved @ +3 dB, with the 100-375 Hz and 1650-2500 Hz range being a simple ramp connecting the shelves. I also don't know if they already applied EQ to correct for magnetic cartridge response.

In fact, the vast majority of the online EQ and speed guides for 78s, both acoustic and electrical, are compilations of unsupported folklore, much of it just copies of what certain audiophiles liked the sound of in the 1970s, '80s and '90s. It doesn't mean they are wrong, but as far as I can tell, very little of it is based on record company literature, analysis of early phonograph circuitry, or measurements and analysis with modern DAW tools. For example, a spectrogram can show exactly what the upper frequency limit is, and the exact pitches of tones in the music. With this information, you can EQ and adjust the speed according to what frequencies the tones should have. I did this and found that despite an online speed guide's claim that a Brunswick 78 should be 80 RPM, playing it at the standard 78.26 RPM yielded A4 notes at the common standard of 440 Hz, whereas at 80 RPM they would be a nonstandard (but not impossible) 450 Hz.

On the other hand, performers made many compromises in order to generate an acceptable sound on the recording, and even with the help of objective analysis, what ultimately sounds most pleasing will always be subjective.(Ref.) No two people will agree on what really sounds best or the most "authentic", nor can a single standard for authenticity really be achieved, given that the phonographs had speed and EQ controls for the listener to adjust the playback to taste.

Reducing and removing noise

Electrical noise

One kind of noise I've had to deal with is electrical "ground loop" hum and/or buzzing which manifests as spikes at multiples of 60 Hz.

Another kind of noise is interference from poorly shielded audio cables and related wiring, including the circuitry inside your computer; it manifests as random whines, clicks, buzzes that correspond to things happening with your hard drive or on your screen, or even as the sound of an analog radio station because the cabling is creating an antenna.

It is better to try to eliminate all of these kinds of noise in your physical setup rather than with noise reduction features of software; otherwise your attempts to reduce it are likely to damage the music. It can be frustrating to try to figure out the cause of noise, let alone to get rid of it, but things that can help are a different soundcard or RCA cables, adding an "isolator" in your cable modem line (it may interfere with digital services, though), making sure everything is plugged in correctly and well-grounded (turntable ground wires are especially sensitive), or simply ensuring that your computer and audio gear is all plugged into the same household electrical circuit.

When recording from tape, I don't worry about the extra noise coming from the player's circuitry or the background noise caused by the magnetic particles on the tape. Sometimes, though, there will be an annoying ground loop hum recorded on the tape. Sometimes I use my audio editor to reduce or remove this hum by using an FFT filter (a kind of EQ) to emulate a set of notch filters at 60, 120, 180, 240, and 300 Hz. It often takes several tries before I get acceptable results; I have to be careful not to try to make the filters too steep. Often, it's best to just try to reduce the noise so it's not so distracting, as opposed to completely removing it.

Sometimes I try to remove electrical noise using my audio editor's noise filter. You first take a "snapshot" of the background noise and tune some parameters. Again, it requires trial and error and fussing with the controls to get acceptable results. It is tempting to use it to wipe out all the background noise, but it is best to just concentrate on the lower, more audible range, because otherwise you will damage the music or introduce some strange effects in the background noise.

Declick, decrackle, depop

For vinyl and 78s I usually have to do a lot of "declicking", which is using the editor to remove or reduce the audibility of ticks and pops caused by scratches and crud in the record groove. Records in Good condition or better tend to clean up quite well digitally, with very little surface noise. Distortion from groove wear can never be eliminated, though.

Declicking can be very fast and automatic, but can also audibly hurt the music because the declicker can't tell the difference between a noise that's not supposed to be there and musical sounds with similarly spiky waveforms or fast attacks, like the sound of a trumpet or the beginning of a tap of a cymbal. So I tend to declick in a more manual fashion by finding the clicks myself and applying an automatic declicker to just each spot of interest, or sometimes even manually moving samples around or excising unrepairable sections entirely. I have it down to a science, but it takes about 3 times as long as the play time of each song to fully clean up just that one song.

Before doing any declicking, I make sure the channels have no DC offset. My soundcard driver takes care of DC offset automatically (it's something I can toggle in the control panel). I also try to get the channels balanced. Depending on what stylus I use, the left channel often has more bass than the right, so I use EQ to make them more even. (I've been unable to figure out a way to correct it in the turntable setup.)

I've read that declickers work better without extraneous bass, so I also usually roll off the deep bass (30 Hz and below) on an even slope in order to reduce tonearm resonance, rumble, and through-the-floor noise. See #Rumble and tonearm resonance.

Manual declicking

In Audition, I set the waveform view to overlap the channels and use different colors. I also enable the spectrogram view (Shift+D) to see the clicks that produce broadband noise; they're vertical stripes that are easily spotted.

The key to manual declicking is that you are only selecting either

  • just the click or pop (plus or minus a few samples is OK), or
  • just the area in between sharp-attack sounds (e.g. slightly less than one beat or bar at a time)

In other words, you are not letting the declicker see the parts of the music it might erroneously mistake for clicks.

In the waveform, I select the click-containing samples (usually in one channel only, toggled using up/down arrow keys) and then I Auto-Heal the selection (which I mapped to key U) to get a near-perfect fix. Or, I might select their general area and apply my declick settings to it (which I mapped to key H), or where it won't hurt the music, max declick settings (mapped to Shift+H).

My declick presets:

  • mjb medium click removal = threshold 52.9, complexity 32.2
  • mjb basic click removal (H) = threshold 21.4, complexity 32.2
  • mjb max click removal (Shift+H) = threshold 1, complexity 100

For comparison, the standard settings:

  • default = threshold 30, complexity 16
  • light = threshold 93, complexity 20 (I will use this sometimes)
  • medium = threshold 50, complexity 56
  • heavy = threshold 13, complexity 95

Bassier pops are harder to spot, but are easily heard and you eventually get an eye for them at the bottom of the spectrogram. When I find one, I select the "lump" in the waveform plus a fair bit beyond on each side of it. I often adjust the top edge of the selection in the spectrogram view so that only the lower frequencies will be affected. I then apply a volume drop, auto-heal, or both. Of course I go back and listen to the result to make sure it's acceptable.

When the pop is in both channels, it's often out-of-phase, i.e. the lump goes up over the zero line in one channel, and down below it in the other. For little ones it doesn't matter, but the louder ones will repair better if the channels stay in phase. So I select the affected samples in one channel, invert them to match the other channel (usually matching the one that forms a peak rather than a valley) and then Ctrl+U both at once. On rare occasion, I'll select individual samples or spans of a few samples and use the volume control to move them into more favorable positions. (If only they'd just give us a pencil tool like I used to use in Sound Forge!)

For the huge pops and other pops which can't be easily repaired, sometimes some surgery is needed, copying a bit of the waveform from the other channel or from an undamaged section, and pasting or mix-pasting it over the partially repaired pop. In extreme cases, when the pop is in both channels and has no musical content (it's just a bright solid line in the spectrogram), I'll just consider it a skip, meaning the needle got bounced out of the groove, so I'll cut it out completely, shortening the length of the file. These are rare in my own rips, because if the needle is getting tossed and slammed against the groove walls, the problem is likely with the turntable setup and should be fixed there.

When there's just too many minor clicks to clean up manually, I might use Audition's declicker, or Izotope's, with very light settings, then undo/redo (Ctrl+Z, Ctrl+Y) repeatedly to see the waveform differences and make sure it didn't "fix" instruments with sharp attacks. And of course I also listen.

After declicking, if there's grounding hum or other tonal electrical interference that's audible and distracting, I'll use general noise reduction to reduce the noise by 20 dB, but really I prefer not to do this because the problem can be resolved externally. I don't worry about the background whoosh of the vinyl. It's just not worth the risk of trying to remove.

Corrective EQ

"Cartridge loading" is an obscure part of audiophile science, where you try to make sure the cartridge, preamp, and interconnects are electrically "matched". Failure to match them results in some wacky treble EQ. Well, after reading about it, I feel the problem is ultimately about minor frequency response differences in the treble range which are correctable with EQ...and then ruined again by the non-flat response of speakers and room acoustics. All part of the fun of playing vinyl, and with audio in general... perfection will always be elusive. A particularly enlightening article on this topic:

Anyway, vinyl is also sometimes mastered with weak treble or weak bass, for one reason or another. Usually I think of it as being an authentic sound as intended by the mastering engineer, but sometimes it is so extreme I just can't believe it was really what was wanted. Maybe the band got a test pressing and played it on a sound system with screwed up frequency response, and ordered changes to make it sound "right". Maybe someone was high as a kite. Who knows.

I keep a couple of EQ presets handy to help fix the worst cases, maybe 1 in 20 rips:

  • bass boost for vinyl = 10 band graphic EQ: 8192 points, 0 4 3 2 1 0 0 0 0 0
  • bass boost for vinyl (75%) = same as above but 0 3 2.25 1.5 0.75 0 0 0 0 0
  • mjb vinyl treble boost = FFT filter, spline curves, 20 Hz @ 0 dB, 1000 Hz @ 0 dB, 1631.9 Hz @ 4.9 dB, uppermost frequency @ 5.3 dB
  • vinyl treble boost = 30 band graphic EQ: 8192 points, 0 until 3.2k, 0.6 1.7 2.7, 3.7, 4.7, 5.2, 5.7, 6.2, 3.7, 0
  • vinyl treble boost (half) = same as above but 0.3, 0.8, 1.3, 1.8, 2.3, 2.6, 2.8, 3.1, 1.8, 0

I also have a technique for fixing the gradually declining treble. From the beginning to the end of a side, there's sometimes a noticeable decline in treble. It is not how the music is supposed to sound; it is just something they do when mastering vinyl so it doesn't sound terrible as the stylus gets closer to the center of the record. Usually this decline is not noticeable, just a few dB, but on rare occasion it's like 6 dB or more. The way I fix it is to copy the entire side's audio to a new window, then I fade out the original audio (applying the fade to the whole side), apply the end-of-side EQ to the entire copy, fade in the entire copy, copy it, and mix-paste it over the original (100% on both, no crossfade).

EQ curves for 78s

Just a selection of curves I have used with some success.

A curve for bass-heavy acoustic recordings ripped with a moving coil cartridge. 8192 points, -2 dB master gain, sliders at 14 until 250 Hz, then 12, 10, 8, 6, 4, 2, 0, -2, -4, -6, etc.
A curve for weak-bass acoustic recordings ripped with a moving coil cartridge. 8192 points, -6 dB master gain, sliders at 30 28 26 24 etc.
A curve for Victor acoustic recordings ripped with a moving coil cartridge. 8192 points, sliders at 29, 27, 25 etc. to 1 @ 800 Hz. Then a logarithmic curve (this is just approximate): 0, -0.4, -1, -1.8, -2.8, -4, -5.4, -7, -8.8, -10.8, -13, -15.4, -18, -20.8, -23.8
another Victor acoustic curve
This one is for the FFT filter. 20 Hz/-300 dB, 66.6/12.1, 311.2983/4.1659, 4062.7693/0.0512, 8099.3955/-300
A curve for bass-heavy Victor electrical recordings. 8192 points, sliders at 21, 19, 17, etc. to 3 @ 250 Hz, then 2.5, 2, 1.5, 1, 0.6, all the rest at zero.
A curve for weak-bass Victor electrical recordings. 8192 points, sliders at 27, 25, 23, etc. to 3 @ 500 Hz, then 1, then zeroes.
Columbia 78
A curve for Columbia electrical recordings. 60 Hz/15 dB, 100/11, 200/6, 350/3.25, 600/1.5, 1000/0, 1708.7961,-2.4866, 3000/-5.5, 5694.4243/-9.9834, 6791.2739/-12.1927, 8099.3955/-300.
sub-300 Hz ramp
On early electrical recordings without vocals, it is pretty safe to boost the bass by quite a bit. A simple FFT filter with 20 Hz @ +15 dB and 300 Hz & up flat at 0 can be applied once or even twice in a row to bring bass notes up to a reasonable level.
mic resonance reduction
Early electrical recordings use microphones which have significant resonant/overamplified frequency ranges, especially on vocals (or anything up close to the mic). It is debatable whether it should be left alone (after all, it was recorded that way), but I feel it is OK to apply some EQ to take some of the harshness out of the worst cases. For example, on Victrola 1092 I used the spectrogram as a rough guide to reduce 500–1250 Hz by about 6 dB and 2700–3175 Hz by about 8 dB. It really needs even more than that but I didn't want to take away too much from the parts in between the loud vocals. Dynamic EQ would be ideal, but that's for another day.
400N-12.3 a.k.a. AES 1951–1958
Ostensibly for LPs, but also a retroactive recommendation for 1930s–1940s US 78s and apparently reasonable for 1950s 78s when other curves don't seem to work as well. In Audition's EQ, use 8192 points, with the sliders at (from left to right): 23 20.6 18.3 16.1 14 12.2 10.4 8.6 7 5.5 4.1 3 2 1.2 0.5 0 -0.5 -1.2 -3 -4.1 -5.5 -7 -8.6 -10.5 -12.3 -14.1 -16.2 -18 -20.2
bandpass filter
A fair number of 78s have all their signal confined to a certain range, e.g. 80 Hz to 5000 Hz, so as a final step I often cut off everything else using a scientific filter (Butterworth, BandPass, 12th order). I adjust the cutoff frequencies based on what the recording actually contains, as visible in Audition's spectrogram.
I EQ and filter on a per-side basis, so as not to lose anything or have any more noise than necessary, but this naturally leads to inconsistent perceived "brightness" when listening to both sides in a row. For example, Victor 19598 (acoustic recording) has higher harmonics on side A than on side B. Side B is pretty sharply cut off at 3250 Hz but side A has some going up to 3600, sometimes much higher, so I EQ'd accordingly. Victor 20241 had nothing above 4 kHz on side A, but had harmonics up to 5 kHz on side B. They were recorded 1 day apart in the same studio by the same band. I guess they were just seated differently relative to the microphones? I'm at a loss to explain why it is this way.

Wondering what the pros do? Well, the first disc of the RCA Victor 80th Anniversary CD box set from 1997 has transfers of 78s from 1917 to 1929 on it. Each track appears to have been EQ'd individually based on its HF content. Many tracks appear to have had a kind of inverse dynamic EQ applied to them, which is really great (after declicking), as it works like noise reduction. I also noticed that a couple of the tracks have that weird sweeping signal in them, so it's not just my records. And every track has noise-shaped dither above 16 kHz.

EQ curves for vinyl

I've used these with limited success. It really is not that different from RIAA though.

Columbia LP 1948–1955
This is for Columbia vinyl records before the RIAA curve was in use. FFT filter, 20 Hz/15 dB, 30/14, 50/13.5, 100/11.9, 200/8.4, 500/2.9, 700/0, 1000/-1.3, 2000/-4.2, 5000/-10.2, 10000/-16.1, 13800/-28.3, stay at -28.3 after that.
NAB 1949–?
Another pre-RIAA curve. FFT filter, 20/-300, 50/10, 500/3, 700/0, 10000/-16, 16000/-300.


I used to normalize each track on a single or EP to have 0 dBFS peaks, thinking it didn't make much difference, and before I learned about "inter-sample overs" and (more importantly) ReplayGain.

Around 2004, I realized that it's normal for different sides of a single or EP to be mastered at different overall volume levels. So now I try to preserve the relative volume levels of the tracks, never adjusting the volume of one track without adjusting all the other tracks on the same side by the same amount. Basically it means treating each side as a separate "album", and that goes for ReplayGain scanning, too.

As for overall volume, I generally aim for whatever will produce a ReplayGain value within a couple of dB of zero.

Fixing wrong-speed recordings

Sometimes someone (possibly me) will accidentally record a 33 RPM side at 45 RPM, or vice-versa. The best way to fix this is to re-record it! But you can fix it digitally in Audition, too.

Speed up 33 to 45

If you are speeding up the rip (it was recorded at 33, but you want to pretend you recorded it at 45), I think the best way in Audition is to do this: Edit > Interpret Sample Rate ... multiply current sample rate by 1.35, e.g. 44100 becomes 59535. Then do File > Save As and under Sample Type choose the original sample rate (also, Advanced: Quality 100%, [X] Pre/Post Filter), and leave the bit depth Same As Source, Dithering Disabled.

Another option, as mentioned in the section above on 78s, is to use a lower sample rate when recording. For example, divide 44100 by 1.35 to get 32667. Tell the software to record at that rate while you play the record at 33. Or, record at 44100 and then resample to 32667. Either way, you can then have the software interpret the sample rate as 44100. It will be as if you sampled at 44100 and played it at 45.

Slow down from 45 to 33

If you are slowing it down from 45 to 33, then the method depends on the original sample rate.

If the original sample rate is 96000 or 192000, you can use the same method as above, but divide instead of multiply. For example: Edit > Interpret Sample Rate ... for 96000 you enter 71111.111111111111111. Then you must reduce the sample rate to 44100 or 48000 when saving.

This doesn't work as well when the original sample rate is low, because when you interpret e.g. 44100 as 32666.66666666667, you lose frequencies above ~16333 Hz, which will make an audible difference. (However, you may be OK with it if you know the audio has no legitimate harmonics above 16 kHz.) So for slowing down a recording that's at a 44100 or 48000 sample rate, you can use an effect:

Effects > Time and Pitch > Stretch and Pitch (process)

  • Algorithm: iZotope Radius, Precision: High
  • Stretch and Pitch: Stretch 135%, [X] Lock Stretch and Pitch Shift
  • Advanced: [ ] Solo Instrument or Voice [ ] Preserve Speech Characteristics

However, these effects are not perfect; they can mangle some sounds. Sample rate interpretation is ideal.

One thing I have not yet tried is sample rate interpretation for the lower frequencies, and the effect DSP for the high frequencies which would otherwise be lost. Think it would work?

Rumble and tonearm resonance

*All turntables produce a certain amount of "rumble"—a subsonic bass tone coming from the motor. The tonearm also has a resonant frequency which generates low-frequency noise. In practice, you get some steady tones somewhere below 30 Hz. If you speed up the recording, these tones move up as well, possibly becoming more audible.

That said, on my SL-1200M3D, the noise is concentrated around 8 to 10 Hz, so speeding it up even from 45 to 78 still keeps the noise below 20 Hz.

Rolling off subsonic noise

There is nothing intentionally recorded in the groove under 20 Hz. You can't hear those frequencies, though you might feel them if they are really loud. Few instruments other than pipe organs go below 20 Hz. In fact, most records are mastered with nothing below 30 Hz; it's just noise down there.

When ripping vinyl, then, it is common to discard, or at least roll off, the lowest frequencies. Personally I like to do it because:

  • I have read that subsonic noise interferes with declicking algorithms. (This was in the Sonic Foundry Noise Reduction DX plugins documentation from 1999, so it may be outdated.)
  • Reducing the amount of noise in any frequency band helps both lossy and lossless codecs achieve greater compression.
  • Strong subsonic noise, though inaudible, is essentially the same as having a bad DC offset—there's just less room for peaks and greater risk of clipping—and your amp is just wasting energy pushing your woofer cones to their limits. Ever see your speaker cones moving violently when playing the silent part of a groove?

In Adobe Audition, most of the EQ filters are unable to apply a gentle slope to the frequencies under 20 Hz; they seem to always produce a steep shelf. My understanding is that a steep "brickwall" filter is undesirable, because you could get "ringing" near the cutoff frequency, though at these frequencies it probably doesn't matter. I don't like the idea of adding noise, though. In newer versions of Audition, there's a Scientific Filter which works very well for rolling off the subsonic noise; just set it to Butterworth, HighPass, Cutoff 20 Hz, Transition Bandwidth 10 Hz.

Another option is to use SoX with the highpass 40 option. I experimented a bit with different settings and settled on 40 as the slope start, because at 20 it really was not providing hardly any reduction to the tonearm resonance. A simple batch file:

E:\apps\sox-14-4-2\sox.exe %1 %1-derumbled.wav highpass 40

Adjust the path to sox.exe as needed and invoke it as derumble foo.wav, where foo.wav is the input WAV. It will create a filtered version as foo-derumbled.wav.