Pure Vinyl

I was looking for information on how to connect a turntable to use Pure Vinyl. What do I need?

A detailed guide with recommended components has been moved to another page.

Also see photographs of setups with audio interfaces on our Getting Started with Computer Audio page.

Information on adapters for connecting turntables to audio interfaces (including DIY wiring diagrams) can be found here.

I read on the Internet that Apple Lossless converts the resolution of 24 bit audio to 16 bit before compressing. Is that true?

No.

Incidentally, Apple Lossless is a variant of the FLAC file format, and uses a similar compression algorithm (although the two aren't strictly identical). Both preserve 24 bits of information. Apple lossless also adds the capability of compressing 32 bit float format, and 176.4 / 192 kHz sample rates, which were unsupported in the original FLAC file specification; the standard FLAC format also does not include a decoder for 32 bit format.

Dynamic range is just one factor that determines the quality of digital audio reproduction. Besides the advantage of using 24 bit resolution during the initial recording to provide extra headroom in setting the recording level, there's also the matter of sampling resolution.

A 16 bit digital word resolves loudness levels in steps that are 1/65,536 of the full scale signal. In other words, when an audio signal is digitized, the permissible values of the digitized result only can occur in these discrete (quantized) steps.

Digitizing a vinyl recording with 16 bit resolution forces the audio to be represented by these 1/65,536 digital loudness steps. This process occurs at every digital sample. (The total number of samples representing the waveform is determined by the sample rate, yet another factor affecting the audio quality.)

With 24 bit audio, the resolvable levels are much more closely spaced together. Instead of 65,536 possible loudness levels, there are 16,777,276. In other words, the loudness "steps" in the waveform will be 256 times finer (this isn't the same as "oversampling," which refers to the sample rate). Consequently, audio sampled at 24 bit resolution more closely resembles the original analog signal than if sampled at 16 bit resolution. Note that "32 bit" resolution, sometimes mentioned in this context, doesn't indicate the resolution of the audio interface; the 32 bits just refers to the resolution of the digital stream used internally for processing by the computer (with two very uncommon exceptions, after the introduction of 32 bit digital converters from AKM and ESS in late 2008). Pure Vinyl's internal signal processing path goes even further than that, by using 64 bits (known as double precision floating point) of resolution.

The reason for using 24 bits rather than 16 bits is just to preserve the original signal as faithfully as possible. "High rez" 24 bit 192 kHz digital audio, reproduced from a high quality audio interface, is widely regarded as being indistinguishable from the original analog signal.

It's all about preserving the signal from the vinyl recording as faithfully as possible; and more.

First, more, about the "more:" as has been discussed elsewhere, sampling at a higher rate moves the phase shifts of the antialiasing (ADC) and reconstruction (DAC) filters outside the audible frequency range. It sounds better! But besides the audio quality, there also is the issue of capturing all that's on the LP:

Unlike the Compact Disc (CD) format, vinyl is capable of reproducing frequencies well above the accepted limit of human hearing. While there is debate about whether or not (or how commonly) humans are capable of perceiving frequencies above 20 kHz, the effect of the "brick wall" filter used in CD production, that sharply attenuates frequencies above the 22.05 kHz Nyquist limit, is harmful to the reproduction of audio, by introducing ringing and phase shifts affecting signal frequencies below 20 kHz. Vinyl has no such limitation on the frequency response; just the smooth roll-off imposed by the mechanical properties of the stylus; no brick-wall.

If you don't believe that vinyl "has" anything above 20 kHz, check out this example, illustrating why it's worthwhile to record / archive the audio at the highest possible sample rate:

Click above image for vinyl frequency response analysis, and comparison with CD "brickwall" 20 kHz response.

The high frequency tails in the audio recording shown above require a sample rate of at least 192 kHz for capture. In keeping with the notion of the definition of archival as being faithful to the original, this information would have been lost if the recording were transferred at a lower sample rate.

Finally, the RIAA compensation filter of Pure Vinyl functions more effectively at higher sample rates.

Accurate emulation of an analog RIAA filter cannot be realized with a general-purpose graphic or parametric equalizer plug-in through a so-called "RIAA filter setting."

Even in the days before digital audio, using a graphic or parametric EQ to attempt to emulate the RIAA curve would have been unthinkable. The reason is that the state-variable, notch or bandpass types of filters used in graphic or parametric equalizers (and FIR, which stands for Finite Impulse Response type filters, used in digital equalizers), while extremely flexible for general audio processing, have incorrect phase and amplitude response profiles for this purpose.

Instead, Pure Vinyl uses first-order digital IIR (Infinite Impulse Response) filters. This particular class of filter is noted for behaving identically to its analog, first order (resistor - capacitor) minimum phase counterpart, while being unburdened by the well-known limitations of physical resistors and capacitors. Pure Vinyl's 64 bit first-order filters for the standard RIAA curve breakpoints were designed by computer modeling as a first step, and then carefully hand-tuned, tested and optimized to match the standard RIAA response, at a variety of sample rates, and finally, evaluated through extensive listening tests, in a process that began years before Pure Vinyl version 1.0 was publicly released.

If you try a parametric or graphic equalizer approach to RIAA filtering, don't be surprised if the results prove to be unsatisfactory, and likewise, such a test shouldn't be used to draw any conclusions regarding the level of audio quality attainable with properly implemented software RIAA filtering.

Pure Vinyl RIAA FIlter Advantages

Audio interfaces such as the Prism Sound Orpheus / Metric Halo have onboard signal processing with an option for an "RIAA filter." However, there are many significant advantages to using Pure Vinyl's RIAA correction, compared to using the RIAA curve in such hardware (or a conventional phono stage):

1. The RIAA filter in Pure Vinyl is minimum-phase, the same as a conventional phono preamplifier. Unilke a conventional phono preamplifier, there is no drift in performance due to component temperature coefficients, no interchannel amplitude or phase variation due to component tolerances, and no added distortion from the RIAA correction.

2. There is an adjustable (frequency and slope) rumble filter which is invaluable for vinyl playback, to remove low frequency information which reduces the headroom in recordings and in the power amplifier.

3. If archiving / transferring to digital, you aren't locked into having a pre-corrected recording. The curve can be applied any time later. You also can produce copies using Pure Vinyl with the RIAA curve applied, in a choice of formats (including with sample rate conversion) so one need not rely on Pure Vinyl for playback.

4. The RIAA filter in Pure Vinyl is calculated with 64 bits of precision. Hardware DSP chips are limited to 32, 40 or 48 bits (depending on the chip). While the signal is in the digital domain, one also can use Pure Vinyl's crossover feature (also calculated at 64 bits of precision), optional high quality 64 bit sample rate conversion, or signal processing or signal analysis (AudioUnit) plug-ins.

5. Besides the standard RIAA correction curve, Pure Vinyl includes over 60 alternate correction curves for use with antique recordings.

6. You don't have to be nearly as careful in setting the recording levels. The RIAA filter actually increases the effective resolution / word length (an AES Convention "white paper" on this topic can be downloaded from this website). This is used to advantage, because it can be integrated with the filter (during playback or optional sample rate conversion of the "flat" recording) if any normalization gain is needed. On the other hand, if recording directly to a 24 bit file with the RIAA curve already applied, any normalization done afterwards will only increase the distortion in the recording.

7. There will be additional options added to Pure Vinyl to take advantage of the characteristics of the "flat" transfer. For instance, pop and click removal is much more effective when used with the "flat" recording, because pops and clicks will not have been altered by the RIAA filter of a conventional phono preamplifier. On the LP the music has been processed with RIAA pre-emphasis but pops and clicks, arising from physical defects in the vinyl or contamination of the LP surface, have not. The RIAA filter acts as a pulse stretcher on the fast rising pop / click impulse, making impulse noise removal more complicated than it should be.

Because of these reasons, it should be apparent why it's much better to use the RIAA correction in Pure Vinyl instead of a plug-in filter or signal processing option provided in your audio interface (or a conventional phono stage)...

That's incorrect, and is a misconception based on a simplified view of the software-based RIAA correction process.

This subject was the focus of a technical "white paper" (available on the download page) presented at the October, 2007 123rd Audio Engineering Society Convention in New York. The slides from the presentation also are available for download.

The paper (#P4-6, Filter Reconstruction and Program Material Characteristics Mitigating Word Length Loss in Digital Signal Processing-Based Compensation Curves Used for Playback of Analog Recordings, http://www.aes.org/events/123/papers/session.cfm?code=P4) proves that software-based correction doesn't compromise audio quality.

In fact, converting the amplified analog vinyl signal to digital with the RIAA treble emphasis intact, followed by applying the correction curve in software enhances the available digital resolution in the midrange and treble, where human hearing is at its most sensitive, compared to digital conversion after RIAA correction. Pure Vinyl performs all internal signal processing at 64 bit resolution, which preserves the benefit of using the pre-emphasized signal (and, as a bonus, no distortion is added). Openness, clarity and definition in the midrange and treble are improved, compared to doing a digital conversion after the signal has been corrected with a conventional phono preamplifier.

More importantly, our Pure Vinyl users have reported (both via the Internet and in person, when visiting our demonstrations at audiophile shows like the Rocky Mountain Audio Fest) that combining software RIAA correction with a high resolution (24 bit / 192 kHz) audio interface (and a quality gain stage) provides noticeably improved "live" vinyl playback, compared to a conventional (including high-end) phonograph preamplifier.

Incidentally, a practical avenue to implementing software RIAA correction as a viable "live" vinyl playback system has only become available in the last several years. The key enabling technology was the introduction of high quality, high resolution audio interfaces, coupled with the availability of powerful, inexpensive computer hardware.

While the iMic is a remarkable device at its price point, there naturally are limits to what can be expected from the performance of a combination microphone preamp and audio interface retailing for approximately $40. (If that were incorrect, then companies selling expensive microphone preamps and audio interfaces, running into the price range of thousands of dollars each, would go out of business. That hasn't happened.)

It's unreasonable to expect to achieve high quality vinyl playback from even a conventional phonograph preamplifier retailing for $40 (they do exist). Likewise, the many engineering hurdles to be met when creating an audio interface that achieves truly high-resolution digital audio reproduction simply cannot be overcome at that price point. If, after listening to your recordings, you find that the iMic isn't delivering the sound quality you expect (as in low noise, dynamic, crystal clear detailed sound that also doesn't cause "listening fatigue" when played at louder levels), then you should strongly consider upgrading your audio interface. The enjoyment of recorded music, whether from vinyl or other sources, always is enhanced by using better quality equipment. And if you're going to invest a lot of time into making format conversions of your vinyl, don't you want to have the best quality possible?

This also extends to the software. Pure Vinyl is designed (leveraging our more than decades-long, in-house expertise in developing audio analysis and signal processing software) from the bottom up to painstakingly preserve all of the sound quality captured by your audio interface. This goal can't be accomplished without the proper care and attention to detail.*

Here are many features of Pure Vinyl not found together in other software:

• Pristine, 64 bit double-precision floating point internal signal path for no added distortion
• Built-in music server for automated playback of music tracks from your iTunes music library
• 64 bit internal precision upsampler for CD quality audio tracks, play at up to 192 kHz 24 bit through high quality audio DACs
• Built-in two-way crossover (can be used for subwoofer biamping), with adjustable time alignment, slope and frequency
• Record to choice of file formats at 44.1/48/88.2/96/176.4/192 kHz, 16/24/32 bit, optional Apple Lossless Compression
• 64 bit sample rate converter (SRC) for creating CD format audio from high-resolution source file, exclusive Channel D design** (see this page for comparisons with other SRCs)
• 64 bit SRC can be used to mirror to a second, CD format file, while simultaneously recording high-resolution audio
• Adjustable, noise shaped dithering for SRC / conversion to 16 bit format
• 64 bit RIAA vinyl compensation curve, for live monitoring or file playback, accurate within a small fraction of a dB
• Library with choice of dozens of customizable correction curves, for playing pre - 1955 recordings
• Adjustable Noise Gate: automatically pauses recording and locks when music stops (such as stylus lift)
• Precision signal level metering with RMS, RMS balance, peak, peak histogram and peak hold
• Controls for channel swap, mono, polarity inversion, precision balance trim
• Unique*** "scratch - style" grab and spin editor uses "vinyl" images rendered from the audio
• Adjustable (frequency, slope) high-pass "rumble" filter
• ZEPHRR® zero-phase rumble reduction filter for track rendering, slope up to 96 dB / octave
• Nondestructive track editor and pop and click editor, with automatic peak detection and normalization
• Split recording into tracks for making CD compilations, iTunes, etc. digital jukebox / iPod
• Fun, animated spinning "virtual vinyl" image during playback re-creates the experience of seeing that spiral-scratch disc spin
• Low Latency NetSend feature for distributing high-resolution (up to 96 kHz / 24 bit) audio to other Mac computers

Note: all updates to Pure Vinyl have been provided free of charge, even to those who originally purchased an activation code for Version 1.0.

*Everything affecting the audio quality of Pure Vinyl has been carefully designed and validated; we didn't simply "bolt together" generic pieces of audio software from other sources.

**Other software running on the Mac may use Apple's QuickTime routines for sample rate conversion (SRC) / pitch alteration, etc. While that's easier on the software developer that chooses to use QuickTime, the quality of QuickTime's audio conversion routines have been regarded unfavorably, even in Apple - sanctioned audio forums. The AudioUnit SRCs available in Mac OS X 10.5 Leopard provide an improvement in quality over QuickTime; see tests of SRC software on our site and elsewhere. Despite this, we still use our own well-tested 64 bit SRC design.

***U.K. Patent #2454106. U.S. patent pending.

DSD is a digital format which has enjoyed lots of attention lately as a superb medium for Internet distribution of high resolution digital recordings, especially reissues sourced from master tapes. So at first glance, DSD would seem to be an ideal format for transferring / archiving vinyl to digital. Unfortunately, this is not the case, especially if the goal is the highest quality digital transfer (or playback).

First, using Pure Vinyl's RIAA correction in software will be out of the question, because (for reasons explained below) recording to DSD will depend on using a conventional phono stage (with analog RIAA correction). That, in itself, presents a severe limitation and will compromise the performance you would otherwise obtain (see topics above describing the advantages of applying RIAA correction in software, rather than in the analog domain).

Why not record to DSD with a "flat" phono stage or preamplifier, and then apply the RIAA correction to the DSD audio, same as with PCM (24 bit / 192 kHz) hardware? Unfortunately, this is not possible with a native DSD audio stream. Mixing, EQ / RIAA correction, normalization, impulse noise removal or other operations are impossible without first converting the DSD audio to standard PCM. (Even the extremely expensive Sonoma DSD audio workstation must use an intermediate DSD to PCM conversion step for mixing operations.)

Avoiding a DSD to PCM conversion step would require using a conventional analog RIAA phono stage and accepting the DSD recording as final. Post normalization, EQ, RIAA correction, impulse removal, etc. are not allowable without PCM conversion; and performing a DSD to PCM conversion would eliminate any putative reason for recording to DSD in the first place.

Accordingly, it's not worth abandoning the ability to perform many very useful signal processing operations just for the sake of using DSD for recording. (This discussion doesn't even consider the significant limitation of the 1-bit delta sigma modulator of DSD recording hardware compared to modern PCM, which uses multi-bit delta sigma modulation offering inherently better quality than 1-bit DSD; or losing the ability to use many of Pure Vinyl's useful automation and time saving features which also would require DSD to PCM conversion at some point: to be able to monitor signal levels for stylus drop triggering, etc.)

Considering these significant issues, the highest quality digital transfers of LPs (and LP playback using software RIAA correction) will be accomplished by employing PCM. DSD has its place as a high quality distribution medium for digital recordings. However, for vinyl, DSD is at a disadvantage compared to modern (even including inexpensive) devices using 24 / 192 PCM digital conversion.

- To manually edit tracks by playing the record, etc. disable SNAPTO. Then, track markers will "stick" where you have placed them.

- If using the track list feature, when entering the track times, instead of entering e.g., 5:12, enter a milliseconds value as well, 5:12.000. This will disable the auto locator for that track and place the marker at exactly that spot.

To edit existing track markers already created with the Auto track locator (or created manually):

1. Disable the ADD/SPLIT and SNAPTO buttons if they are enabled.

2. Select the track whose cue-in you want to edit.

3. If the cue-in is too "late" (comes after the desired track break), move the stylus to before the cue-in.

4. Press play, and stop play when the playhead reaches the spot you want.

5. Click LINK/COPY and click the Copy button.

- The above can be used to edit the Cue Out as well (click the OUT button).

- If using the waveform display to locate the cue-in, omit Steps 3 and 4 above. When you have located the cue-in, proceed with Step 5.

- If you want to move the track marker on the fly while playing, don't stop play (step 4). Click LINK/COPY. At the instant when the playback reaches the desired place, click Copy.

- If you want to move the cue-out of the previous track to correspond to the new cue-in of the track above, select the previous track, click LINK/COPY and click Link. The resulting edit will be gapless with respect to the two tracks.

Be sure to click the SAVE button to save any changes.

This can occur sometimes when seeking near the beginning of a Losslessly Compressed recording. Use the left / right keyboard arrow keys instead of manually spinning the vinyl to set the lead-in for the first track. The keys will move (when in Mark-IN or Mark-OUT mode) the playback position by one revolution of the disc (1.8 seconds for a 33 1/3 RPM album), or fine-tuned by depressing the Command or Caps Lock keys. Once you have set the lead-in coarsely (click M-IN in the Edit panel) the position can be fine-tuned by spinning the disc manually.

Confirm that your audio interface is functioning properly.

First, Quit Pure Vinyl. Open Apple's Audio MIDI Setup application (in /Applications/Utilities) and insure that your audio interface is selected as both the Default Input and Default Output device. Try playing some music from iTunes. If the audio interface is functioning correctly, you should hear playback from iTunes.

If the interface doesn't appear in Audio MIDI Setup, there is a software or hardware fault with the interface that must be corrected before proceeding. Consult the documentation for your audio interface for assistance.

Pure Vinyl Recorder uses a large sound input buffer that causes the "latency" you are experiencing. A large sound buffer serves as extra insurance from skips in the sound as the recording is saved to disk. It is not cause for concern and doesn't harm the recording quality.

This amount of latency might be a problem for multichannel studio recording (overdubbing), but doesn't cause problems with two-channel audio transferred from pre-recorded sources (or when recording live sources without needing overdubbing).

A side benefit of the latency is that it will decrease the susceptibility of your sound system to audio feedback between the loudspeakers and turntable, if monitoring a vinyl recording at a high listening volume.

Use the "First or currently playing" option in the iTunes control dialog in Pure Vinyl, and sort the playlist by track number in iTunes, using the title bar of the list in iTunes.

If you want to manually select the next song to play within a playlist, just double-click the song in the playlist before the one you want to be the "next" track to play. Then, Pure Vinyl will increment your choice and pull the next track from iTunes. If using this method, be careful to first reduce the playback volume to zero in iTunes (with the volume slider); otherwise, iTunes will then play the music (possibly quite loudly) directly through your audio interface.

The Preview Release of Pure Vinyl 3 has complete iTunes jukebox support.

From your email you've indicated that you're running Mac OS X 10.4.10 on an Intel Processor Mac. There is a known problem with this OS X update on Intel processor Macs, causing audio devices to stop working. This problem isn't caused by Pure Vinyl.

The solution is to update to Mac OS X 10.4.11 or later, or to download and install the OS X 10.4.10 v1.1 Combo Update from Apple from their OS support downloads page:

http://www.apple.com/support/downloads/macosx10410comboupdatev11intel.html