Thursday, January 23, 2014
Manley Debuts CORE Reference Channel Strip
Manley Labs today announced the CORE, an analog channel strip.
Manley Labs announced the CORE, an analog channel strip. CORE is an innovative and affordable mic preamplifier, compressor, equalizer, and limiter combo-unit that combines the greatest hits of the Manley product line with fresh technology.
The intuitive design incorporates musical and forgiving circuitry that allows the user to concentrate on performance rather than be lost in a sea of knobs. No other channel strip at this price point offers higher headroom or higher end sound than the CORE, which is, like all Manley products, handcrafted in Southern California.
“As more musicians are contributing to a project remotely, coached by the recording engineer over the telephone, we saw the need to provide an affordable and easy-to-use, excellent sounding recording channel for these guys. They aren’t engineers, they are musicians!” commented EveAnna Manley, president of Manley Labs. “The CORE is feature-laden without being confusing. Its whole purpose is to give the musician the tools he needs to turn in a great sounding track.”
Front panel controls include 48V Phantom power switch, 120Hz High Pass Filter switch, Phase Invert switch. Input Attenuator (Variable Pad), Mic Pre Selectable Gain 40 dB or 60 dB (Total Gain >70 dB) and Line Amp Selectable Gain 20 dB or 40 dB,
The Mic-Line preamplifier features a hand-wound Manley IRON® input transformer with nickel laminations in a mu-metal can, Class A tube amplifying stage circuit topology (similar to the Manley VOXBOX® and Manley Dual Mono & Mono Microphone Preamplifiers), all-triode high voltage vacuum tube circuit and regulated 300 Volt B+ supply.
A quarter-inch direct input is similar to the DI in the Manley SLAM! with all-discrete solid-state circuit and 10 Meg Ohm input impedance (ideal for guitars, bass, keyboards, etc.).
The on-board compressor utilizes the ELOP technology also found in the Manley VOXBOX and is placed before the mic preamp making it virtually impossible to clip. It offers a ratio of 3:1, continuously variable Attack, Release, and Threshold controls and a silent bypass switch.
The equalizer has low and high Baxandall shelves (80 Hz and 12 kHz) with ± 12dB range and a sweepable midrange bell EQ (100 Hz – 1 kHz) or (1–10 kHz) with ±10dB range.
A fast attack FET brickwall limiter offers continuously variable threshold and release controls, a Peak Limit LED indicator and 10dB range output gain control.
A large illuminated analog display provides a 3-way meter select to read Compressor gain reduction, Mic Preamp Output level, and Main Output audio levels.
Inputs and outputs include balanced XLR mic inputs and line inputs, a front panel direct instrument 1/4” input, insert point between Mic Preamp/Compressor and EQ/Limiter via 1/4” TRS jack, balanced XLR direct output (after Preamp/Compressor section) and balanced XLR main output.
Availability and Pricing: The Manley Core is available Q2 2014 at a MSRP of US $2250.00.
Posted by Julie Clark on 01/23 at 06:41 AM
Tuesday, January 21, 2014
Rupert Neve Designs Launching 551 Inductor EQ At 2014 NAMM Show
Includes three bands of EQ inspired by Neve’s most prized vintage designs
At the 2014 NAMM Show, Rupert Neve Designs is launching the 551 Inductor EQ, the first and only equalizer for the 500-series designed by Rupert Neve.
The 551 includes three bands of EQ inspired by Neve’s most prized vintage designs, along with custom-wound inductors, transformers and class-A gain blocks, it brings the thick, powerful lows and sweet highs of Rupert’s classics to the 500-series format.
The 551 echoes Neve’s classic 3-band EQ feature set, with a custom inductor, switched frequencies and a HPF. Traditional transformer-coupled inputs and outputs designed specifically for the 500-series are used for both technical performance reasons and optimum musical reproduction.
The 551 will begin shipping in late January 2014 with a U.S. list price of $950.
The 3-band, custom-tapped inductor EQ on the 551 was inspired by RND’s favorite portions of Rupert’s vintage EQ designs. The low frequency band is designed to produce a creamy, resonant bass response – however, unlike the vintage modules that inspired it, the LF band on the 551 can be used as either a shelf or a peak filter, adding punch, dimension, and control to your low end.
The 551’s inductor midrange band is ideal for sweetening vocals and instruments while bringing them forward in a mix, and its proportional “Q” response makes it well-suited for minimizing problematic frequencies.
The high-frequency band is a hybrid vintage/modern design, blending inductor circuitry with capacitor-based topologies to achieve vintage tones with enhanced control.
The high-pass filter is a 12 dB/octave design with a fixed 80 Hz frequency, and can be used in tandem with the low frequency EQ to add low-end presence without clouding the source material.
As Neve originally intended with his most prized classic designs, each EQ section uses low-feedback, class-A discrete electronics to prevent low-level artifacts and harshness from detracting from the tonal shaping.
However, the updated EQ circuit of the 551 is a decidedly modern design using techniques and components that were simply not available 35 years ago, and should not be considered a “clone”.
Both the high and low band can be switched from shelf to peak curves and offer 15 dB of boost or cut. The high band can be switched from 8 kHz to 16 kHz, and the low band can be selected at 35 Hz, 60 Hz, 100 Hz or 220 Hz.
The inductor-based mid band offers six center frequencies; 200 Hz, 350 Hz, 700 Hz, 1.5 kHz, 3 kHz and 6 kHz. The mid band also has a “Mid Hi Q” switch to narrow the bandwidth (increase the Q) of the filter.
“While creating functional 500-series modules is relatively simple,” Neve states, “designing those modules to equal their non-500-series counterparts with the current, voltage and space restraints is quite challenging. In creating our own 500-Series modules, we experimented with a number of different transformer and circuit designs to achieve the same presence and sweetness found in the Portico Series of modules.
“The result of these efforts is that outside of the slightly lowered headroom, our 500-series modules are nearly indistinguishable from standard Portico Series modules, and are perfectly suited for studios of the highest caliber.”
Another element unique to the 551 is the custom-wound inductor in the EQ circuitry. Inductors are wires wound around a coil that provide a form of frequency-dependent resistance. When they saturate, they bring out beautifully musical harmonics that give tracks the smooth, polished sound that has made Neve’s consoles and equipment so desirable for over 50 years.
With an extreme attention to detail towards variables like the winding and core materials in relation to the surrounding EQ circuitry, the 551’s custom inductor helps the EQ capture the vitality of Neve’s vintage modules, while still retaining its own sonic signature.
Rupert Neve Designs
Posted by Keith Clark on 01/21 at 05:51 AM
Monday, January 20, 2014
Church Sound: The Basics Of Mixing Console Channel Inserts
One of the features often found on the rear panel of a mixing console is the channel insert. The insert serves simultaneously as both an input and an output for either a single channel or for some other signal path, such as a submix or main output bus.
It is a point in the signal path at which the signal can be detoured — sent out of the mixer — and then returned to its normally scheduled programming, creating what is called an effects loop. In other words, it allows you to “insert” an outboard device into the signal path.
On many mixers, a single 1/4-inch three-conductor jack provides connections for both an input and an output. What would you do with such a strange jack?
1. Apply effects to a channel or submix. Because an insert is both an input and an output, you can route the signal from the channel out to a reverb, compressor, limiter, etc., and then back into the channel. You might send the signal to a noise gate to automatically “turn off” a mic when it’s not in use.
Reducing the number of mics that are on, or “open”, reduces the risk of feedback and improves your signal-to-noise ratio.
2. Use it as a direct output, like a post-mic preamp, but pre low-cut filter, mute, EQ, fader, etc. Just because you’re sending something out doesn’t mean you have to bring it back. You can use each insert to send a “direct out” signal to a line-level input of a tape recorder, or to another mixer for a broadcast or recording feed.
At the mixer end of your direct out cable, you’ll want a standard 1/4-inch mono (or TS, tip/sleeve) phone plug. Push the phone plug part way into the insert jack, just to the first click. This will route the direct out signal via the cable, without interrupting the signal flow in the mixer.
If you insert the plug all the way to the second click, you will still get a direct out signal, but the signal in the channel will be interrupted at that point — removed from the mix.
3. Insert a signal through a “Y” cable — using the insert as both a direct out and an effects loop. As an alternate approach, create your effects loop as described earlier, then insert a “Y” adapter after the processor to affect (compress, for example) both the direct out and the individual channel in the mix.
A good application for this might be to compress a pastor’s lapel mic or a pulpit mic, in both the house mix and a recording or broadcast.
Whether you use them as part of your normal setup every week, or just to solve an occasional routing problem, inserts add tremendously to the versatility of your mixing console.
Article provided by Mackie.
Harrison Consoles Unveils 950mx Analog Console (Includes Video)
For facilities that need an analog monitoring, mixing, and summing solution when working with a DAW
Harrison Consoles has introduced the 950mx analog console, intended for facilities that need an analog monitoring, mixing, and summing solution when working with a DAW.
The 950mx provides large-format console sound and construction while forgoing the expensive multitrack buses and inline monitoring features that have become less necessary with modern DAW workflows. It can act as a centerpiece for a commercial recording studio, live 2-track recording rig, or project studio.
Harrison has been building consoles since 1975, and the 950mx incorporates design elements found across a wide range of console models such as the 32 Series, MR Series (2, 3, 4, 5 and 7), SeriesTen, SeriesTwelve, MPC and LPC.
The 950mx offers a robust ground plane, gold module connectors, gold-plated switches, conductive plastic knobs, and fully-differential balanced I/O at every point. The summing buses are carried via PCBs, not ribbon cables. A custom-designed linear power supply provides rock-steady voltage for clean sound, robust EQs, and generous headroom.
All 950mx mono input modules now feature individually switchable, sweepable high-and-low pass filters, and 3-band sweepable EQ - featuring the same circuitry as the original Harrison 32-series consoles.
Mono input channels also feature switchable inserts, a mic pre (switchable to a line input), four aux sends, 100mm P+G faders, two mix bus assignments, and a post-fader direct output.
Stereo input modules feature switchable high-and-low pass filters, 3-band tone controls, balance, channel reverse, mono-sum, input trim, four aux sends, 100mm P+G faders, and two mix bus assignments. Another user-requested feature was the addition of an alternate speaker output.
The mono and stereo modules of the 950m and 950mx are interchangable, and, 950m users can arrange to have their mono channels updated to the reflect the new features of the 950mx.
The 950mx console also includes discrete mix buses, each offering a different flavor at mixdown. Mix Bus 1 features a transformer-balanced output, while Mix Bus 2 is electronically-balanced. The buses can be used separately or may be summed to achieve effects like parallel compression.
The 24-module version is now available in a stand-alone format with optional legs. For customers with existing analog gear, Harrison has designed a matching line of studio furniture to complement the 950mx console. These pieces include a “sidecar” which provides 10 rack spaces at the desk surface, and a 16-space standalone rack. Both pieces use the same laser-cut and powder-coated aluminum framing as the 950mx - providing a professional and cohesive look for any 950mx user.
Unit Audio Debuts New 16-Channel Analog Summing Mixer With Panning
Addition of two pan switches allows placement of channels 1 and 2 in monaural (center), or hard left (ch 1) or hard right (ch 2)
Unit Audio has introduced the Unit 16 x 2, the latest addition to the company’s line of passive analog summing mixers.
It offers the same panning flexibility as the 8 x 2 Micro-Unit with the addition of two pan switches that allows placement of channels 1 and 2 in monaural (center), or hard left (ch 1) or hard right (ch 2).
Unit Audio analog summing mixers are designed to add back some of the “sparkle and punch” of analog recording that can go missing in a purely “in the box” mix. (Sound samples are available on the company website here.)
“With modern DAW software, mixing within the computer has resulted in some great sounding recordings, but I have long been intrigued by the concept of analog summing to get my DAW mixes out of the box,” explains Unit Audio design engineer Terry Auger. “I was not prepared to pay $800 or more to find out for myself, so I engineered and built my own.
“Then to test the theory,” he continues, “I set out to see if there was any difference in the mixed sound. Much to my amazement and pleasure, I did notice a subtle but very pleasing difference in the stereo separation and placement of the instruments compared to my “in the box” mixes. If you’re mixing on an analog console or through outboard analog gear, you have no use for these mixers, but if all of your mixing is done in the box, you will be surprised at the difference they make”
Unit Audio mixers are built hand-wired, point-to-point in Nashville, TN, using top-quality components like Neutrik connectors and Xicon resistors.
Pricing for the new Unit 16 x 2 starts at $299 (plus shipping), with further information, options and purchase available here.
Thursday, January 16, 2014
DAD AX32 From NTP Technology Now Supports 48 Analog Channels
The DAD AX32 is now available with version 1.3 firmware which accommodates up to a total of 48 channels of analogue inputs or analogue outputs or a combination thereof.
NTP Technology announces a major addition to the capabilities of its DAD AX32 ultra-high-quality digital/analogue/digital converter.
The AX32 is now available with version 1.3 firmware which accommodates up to a total of 48 channels of analogue inputs or analogue outputs or a combination thereof.
As a result, the AX32 can have 48 microphone preamplifier and converter inputs or a combination of 32 input channels and 16 output channels.
A total of six analog cards can be housed in the AX32. Each card slot accepts an eight-channel line-level A/D converter, eight-channel microphone input or eight-channel D/A converter with analogue line output. The cards can be combined in any permutation.
Version 1.3 also adds the ability to control the sampling rate of the IP audio interface powered by Dante (TM), and full support for redundant IP audio connection plus a word clock output feature supporting all the sample rate frequencies (44.1 to 192 kilohertz) or just the base sample rate of 44.1 or 48 kilohertz.
All the functionality is controlled from the DADman 4.0 with new comprehensive features for channel strip management of all the channels, as well as an improved router matrix configuration.
Existing AX32 units can be upgraded with the new firmware and DADman control software. New firmware for the Dante Brooklyn II IP Audio card is also available. System configuration is highly flexible and includes IP Layer 3 Ethernet-based audio networking powered by Dante.
Up to 48 full-bandwidth channels can be forwarded via the AX32 along a single Category 5 cable using IP. This ensures flexible and cost-efficient cabling from a recording studio to a control room. The connections can be part of a standard Gigabit Ethernet local area network.
The optional microphone preamplifier incorporates analogue gain control in 0.5 decibel steps as well as digital gain control with 0.1 decibel precision. Dynamic range of analogue to digital and digital to analogue conversion is 126 decibels. The preamplifier’s equivalent noise floor is -132 decibels.
The AX32 is equipped with AES/EBU and MADI inputs and outputs plus a Dante interface for use with compatible third-party products as well as audio workstations running on Apple OS X or Windows platforms.
A versatile interface structure allows the analogue-to-digital, digital-to-digital and digital-to-analogue converters to be assigned to any digital interface, as well as patching between the interfaces on a channel-to-channel basis.
The AX32 can be controlled from the front panel which interacts with four adjacent pushbuttons to allow adjustment of key converter parameters. Full remote control can be accomplished using NTP Technology’s DADman software or via Ethernet.
Existing users of DAD audio converters include Abbey Road Studios, Alchemy Mastering, Bauer Studios, Benny Andersson’s RMV Studio, Classic Sound, CMC Studios, Collegium Records, Danish Radio, DEX Mastering, DPA Microphones, Echopark Studios, Galaxy Studios, Hana Music Montreux, Helsinki Music Centre, Lindberg lyd, Magne Furuholmen, Master Touch, McGill University, Moscow Music Conservatory, NDR Hamburg, NHK, NRK, Opéra de Dijon, QVC shopping channel, Real Sound, Royal Danish Opera House, Royal Opera House London, Sidney Opera House, SK Works, Slovak Philharmonic Orchestra, SoundWorks/Jeff Sheridan, Spanish Radio, St Petersburg Philharmonic Orchestra, Stock Fish Records, Swedish Radio, Telarc International, Timbre Music, Ultimo Productions and the Warsaw National Philharmonic Orchestra.
Posted by Julie Clark on 01/16 at 10:41 AM
Wednesday, January 15, 2014
Tech Tip Of The Day: Tape Transfers
Q: I generally work entirely with DAW in my studio; however, once in a while I get the odd project that causes me to bring out all manner of old gear from storage. I was asked to do some DAT transfers for a local church I’m friendly with as a favor, which I’m always happy to do.
They’ve recorded some worship songs to DAT and want to distribute them to some local residents via (of all things) cassette tape.
What’s my problem? Well…once everything was transferred to my DAW and cleaned up, I started making a transfer to the cassette. However, the playback volume level of the cassette is lower than any store-bought cassette I have lying around.
How can I get the levels up to be comparable to commercially available recordings? All of my meters are as close to red as I can get them without going into the red and causing distortion.
A: Funny how meters don’t always really tell us what something sounds like, isn’t it?
In a word, the answer is compression. Proper use of compression will raise the average sound level and thus raise the perceived volume of sound without making peak levels much higher.
Limiting can be used to raise average volume levels without any increase in peak levels. One has to be careful though, because these processes are often accompanied by side effects that change the sound of the audio, sometimes in undesirable ways.
Engineers cope with this in a variety of ways. Basically you can compress your tracks before an initial A/D conversion, as a process during mixing, or any combination of these.
Normalization, another variation on compression, is another useful process to employ during mixing and/or mastering because it is a relatively easy way to substantially increase the perceived volume of your tracks without too many audible artifacts (depending upon how severe the normalization is).
As for your final step of going to cassette, the problem is still mostly compression. First, don’t be afraid to slam the levels on the cassette. I don’t just mean slightly going into the red. I’m saying, “slam” them. Put your pre-recorded tape in and look at the average levels of the meters, then look at the average levels of your recording. I’m not talking peak levels; I’m saying look at where the meters spend most of their time. Yours is probably lower.
You need to compress (or normalize) until you can get a higher average level without higher peak levels (they will introduce distortion), and may find that you can’t get the average level as high as the prerecorded tape without the tape distorting. This is caused by two factors:
1) Your tape machine isn’t optimized (biased) for the type of tape you are using. Corollary to this, and possible third factor is that your tape just isn’t as good as it needs to be.
2) Your recording has too much very low and/or very high frequency information in it to easily be recorded on tape at high levels. Analog tape is not linear. It doesn’t do as well at very low or high frequencies, especially if noise reduction is being used, and especially if there are big transients with those frequencies.
The solution may be as simple as rolling off some low- and high-frequency content, or you may have to go back to your source and pull some out some of the low bass. There’s no reason, for example, to try to record much of anything above about 12 kHz on a cassette. Most cassette recorders are not going to record it very well (if at all) and it eats up a ton of headroom.
If you get the proper equipment and experiment enough with these factors you will find that you’ll soon be getting 95—99 percent of the level on your tapes as the pros.
For more tech tips go to Sweetwater.com
Friday, January 10, 2014
The 10 Most Frequently Asked Questions About Mastering
1. What is mastering and the role of the mastering engineer?
Mastering is essentially the step of audio production used to prepare mixes for the formats that are used for replication and distribution. It is the culmination of the combined efforts from the producer, musicians, and engineers to realize the musical vision of the artist.
Each stage of the audio production process, from pre-production to mastering, builds on each other and is dependent on the previous process.
Mastering is the last opportunity to make any changes to positively affect the presentation of your music before it evolves from a studio environment to the outside world.
An awareness of the differences between the roles of mixing and mastering engineers should be noted. While the tools may be similar, the perspectives between mixing and mastering are very different. When mixing, the focus is on the internal balance of individually recorded tracks and effects used both sonically and creatively for a single piece of music.
An album cannot be heard in its entirety until the job of a mix engineer is completed. The mastering engineer picks up where the mix engineer leaves off. Mastering is geared toward creating the balance required to make the entire album cohesive. The mastering engineer is most concerned with overall sonic and translation issues.
A mastering engineer works with the client to determine proper spacing between songs and how songs will be ordered on the CD. The flow of an album must appeal to the listener; it should engage them and take them on a musical journey as determined by the artist. Any final edits will be addressed during the mastering process as well.
Finally, the role of the mastering engineer is to provide preparation and quality control of the physical media send to the plant for replication. This includes listening to the premaster CD to verify integrity, along with the more technical aspects such as encoding text, UPC/EAN and ISRC codes, checking for errors within the media and providing any necessary documentation such as a PQ list.
2. Is mastering always necessary?
A writer’s words are not complete until the editor approves them. A painter’s work is not complete until it has been matted and framed. A musician’s work requires the same treatment. Audio production should not be rushed, finished haphazardly or completed “just to get it out there”. A finished product should reflect all of the work of the artist, producers and engineers that carry that vision forward.
Even a “perfect” mix needs mastering to a degree. In this case, you want the mastering to be as transparent as possible so that the original sound is maintained while preparing it for the final media.
As mentioned earlier, it is difficult for a mixing engineer to know how an entire album will sound in its entirety while mixing an individual track. In some cases a given track may be perfect on its own. However, when that track is placed within the context of an album, slight adjustments in level or frequency balance may be required.
Given the amount of music distributed online, an album needs to stand up from start to finish to be noticed in such a competitive market. If the final goal is to create a product that is ready to be played on the radio, distributed online, or sold as a physical product, it should be mastered.
Mastering helps say something about the professionalism of the artist, from the arrangement of certain styles of songs to the volume of the recording to the pacing of the tracks. If an artist is serious about their music, they should make sure that someone with experience signs off on the finished product.
3. What kind of improvements can be expected from mastering?
Mastering can help to achieve the correct balance, volume, and depth for a style of music. It can add clarity and punch to music, giving it more vitality.
The idea behind mastering is that a product will sound better after it is treated by the mastering engineer. The degree with which a mastering engineer can achieve this is dependent on the given mixes. In some cases there may be limitations or compromises that need to be made.
One limitation of mastering is the inability to restore severely distorted material. Distortion in a mix is like corrosion; once present it cannot easily be removed and has permanently destroyed a part of the material.
While mastering can mask the effect of some types of distortion, it is essentially covering blemishes that should be addressed before the mastering stage. A common misconception is that mixes should be as “hot” as possible. With the advent of 24 bit digital technology there is no reason why mixes have to “go into the red.”
Most mastering engineers recommend a cushion of anywhere between -6 to -10 dBFS from peak level to help ensure that clipping does not take place and to allow room for processing. In addition to peak level, the crest factor (peak-to-average ratio) is very important. While dynamic range can always easily be reduced, it is very difficult to undo the effects of over compression or limiting.
If the internal balance of a stereo mix is off, there may be compromises in the sound of the mastered track that will need to be made. For example, if cymbals or a vocal is very sibilant and bright while other parts of the mix are dark, it can be difficult to balance the overall sound in a way that enhances all elements.
In addition to frequency, levels between tracks may also be an issue. If the mastering engineer is given a stereo mix (as is usually the case) specific individual components of the mix cannot be completely isolated and processed separately.
While there are techniques such as de-essing, mid/side processing, equalizing or compressing for a specific imbalance, the results will likely not be as good as with a mix not having these issues and allowing the mastering engineer to address the balance on the whole.
One method of getting around internal balance issues is to provide alternate mixes. Some examples are vocal up/down mixes or mixes where one EQ is favored over another. Another method is supplying the mastering engineer with “stems” or sub mixes of the stereo track.
These might include a separate stereo mix of the vocals or instruments that when summed together are the same as the stereo mix minus any stereo bus processing.
In this case the mastering engineer is placed slightly in the role of a mix engineer and can make adjustments that wouldn’t be possible with a stereo mix alone. Another advantage with using stems is that alternate masters can easily be created such as radio edits, instrumental and vocal-only masters.
Another area where “fixing it in the mix” is better than “fixing it in mastering” is when dealing with the issue of noise. Mute automation on individual tracks should be used where there are noises during sections of a track that are not contributing to the mix.
Some examples are electric guitar hum/buzz on intros, outros, and breaks, bleed from headphones on the vocal track when the vocalist is not singing, drummers laying down their sticks after cymbals have faded but while other instruments are still playing at the end of a track.
4. What are some tips to help ensure the best possible master?
Audio quality can be very subjective. Before hiring a mastering engineer for a project you should have a clear objective on how you would like the finished project to sound.
Communication of these objectives between client and engineer is a key component to the success of a project. The language used to describe the character of audio can be ambiguous. Terms like “brassy,” “fat” and “present” mean different things to different people.
One of the skills of a great mastering engineer is to able to translate this loose terminology into the specific technical processes required to achieve the client’s goals in a non-obtrusive way.
Some mastering engineers find reference tracks from clients to be helpful. Reference tracks can be worth a thousand words, because they serve to demonstrate the sonic objectives of the client.
My personal preference is to receive mixes that are as close as possible to what the finished product should sound like, but with enough leeway to be able to manipulate the sound in order to mold a cohesive album. Some general tips toward achieving this are:
Knowing your room and monitors. If you are using smaller nearfield monitors for mixing, be sure to listen to the mixes on a system that has extended bass to ensure that there are not low end bass problems.
If your monitors or room “color” the sound in any way be sure to compensate to ensure that the mix will translate well on other systems.
Fix track related issues before mastering. Listen for issues like excessive sibilance, uneven or exaggerated frequencies, phase or polarity problems, bad edits, depth and width of the sound field, and the relative levels of instruments and vocals.
I recommend listening to a mix in mono in order to hear if anything disappears or becomes exaggerated as well as listening to the mix at different levels and positions within your room. This can sometimes make an issue more obvious due to a different perspective.
Leave enough of the mix dynamics intact, so that the engineer can make adjustments not only in the overall level but in the punch and clarity of the transients.
Don’t use any processing on the master bus that will interfere with processing that is best performed while mastering. This may include exciters and harmonic enhancers, EQ, normalization and limiting used to achieve a higher overall volume.
Leave the heads and tails of a mix intact so that there is room ambience before the music starts and enough of the music at the end to be able to tailor the fade out. Having a bit extra at the start and end can also be useful so that a “noise profile” can be created for noise reduction systems that use this as a technique for removal of broadband noise.
Use mute and volume automation to remove extraneous noises from the individual tracks. Noises include headphone bleed when the vocalist is not singing, hum from electric guitars during breaks, and a drummer who may lay down his sticks after the cymbals fade at the end of a song, but before the final fade out of other instruments.
5. What should I send to the mastering engineer?
Mixes should be delivered in a format that alters the sound by the least amount. For digital mixes, an uncompressed format (AIFF or WAV) should be used rather than compressed formats like MP3 or AAC.
You should speak to the mastering engineer that you will be working with to verify the formats that they accept.
I recommend staying with the same sample rate used in the original tracks, unless mixing through an external converter. In that case, increasing the sample rate has its benefits. The bit depth should match the one used during the mix session rather than supplying tracks on audio CD where truncation and optionally dithering of the original tracks is applied.
I also prefer that digital mixes be sent as a single stereo interleaved file rather than split stereo files in order to help ensure phase coherence. While a standard when sending analog tape for mastering, reference tones are becoming a lost art with digital.
If mixing through an analog board or to an external device, having an unaltered 1k reference tone (corresponding to 0 VU on the console) can help to identify issues where left and right channels are not calibrated or set properly. If you’re not attending the session, be sure to send all documentation regarding the sample rate, bit depth, format, and a listing of the filename with the full name of the song for each file.
Also note if there are alternate mixes of the same track (e.g. vocal up/down). A listing of the song order is also necessary along with requirements for song spacing and fades if not printed on the original mix. If CD text, UPC/EAN or ISRC codes are to be added to the final CD they must also be included in the listing.
Documentation may include information about your audio chain such as equipment and processing used (particularly if applied on the overall mix), what you feel are some of the enhancements that you would like to hear in each mix, along with any other information that you feel will be useful to the mastering engineer.
6. How much will mastering cost?
Prices vary depending on the profile and experience of the engineer, previous credits, along with studio costs and overhead. Typical rates are based on:
—Flat rate per album usually tiered based on the total number of tracks, sometimes with a total hourly cap.
—Flat rate per track or number of minutes per track.
—An hourly rate that can include additional costs for media due to time spent verifying and listening to the disc.
Some studios may also charge more for attended sessions versus non-attended sessions where the final product is delivered and approved by mail or Internet.
Costs for mastering vary anywhere from $10 per song to $500 per hour for well regarded professionals. Given that mastering is a subjective service-based business, as opposed to a commodity which can more easily be compared, caveat emptor applies.
Assuming that both quality and cost are considerations, set a realistic budget for mastering at the start of your project. Sometimes independent artists will not have anticipated the costs for mastering until a project is completed. This forces them to use lower quality alternatives that are not necessarily best for their project. It’s a good idea to research the studios that will work within your budget. Call them to discuss the details of project and their approach.
In addition to gaining a better understanding of their process you will be getting a feel for the quality of their customer service. Some studios provide a demo of your material to ensure that they meet your expectations; others may charge for this service.
In either case, this is a good way to hear the quality of their work before committing to the cost of an entire album.
7. How much of a role does gear play versus the talents of a mastering engineer?
As the saying goes, “It’s the driver not the car”. A good engineer can work around limitations while a bad engineer will likely produce poor results, great gear or not.
This does not entirely discount the aspect of the gear. Having gear which is made specifically for mastering makes a big difference, not only in the quality of the sound, but in how quickly and easily the engineer can perform his work.
This includes equalizers, compressors and the usual components that most associate with the term “gear” as well as quality converters, monitors, and the room where the mastering engineer works. Any of these can skew the perception of what an engineer hears potentially causing them to make decisions that wouldn’t happen given better accuracy.
There are many hardware and software companies claiming the ability to allow anyone without prior experience to use a particular preset, match frequency curves with references, or use other methods which will allow them to master their own music.
These “cookbook” approaches really miss the point of what the mastering process should be about. This approach cannot replace the skill acquired by an experienced engineer.
The processing performed should bring out the elements of the mix that are most important to each song. This requires both an artistic and technical evaluation.
Using a generic EQ or compressor setting to try to achieve this doesn’t address the individual characteristics of the song that make it unique or the specific problems that it may have in translating those elements.
8. What is the best (fill in the blank) for mastering?
This is a question that is often asked within mastering forums. The simple answer is that there are no “best” or one-size-fits-all solutions. If there were, mastering houses would look more like a chain of department stores with the same type of room, monitors, and gear.
Just as the processing chain used for a particular piece of music will vary according to the character of that track, the hardware and software chosen by an engineer is based on his workflow and tastes.
There are however some common characteristics among mastering studios. The following are what I would consider the universal set of tools ranked in order of importance.
—A discriminating pair of ears. The ability to critically analyze issues that will interfere with the enjoyment and translation of a piece of music is the most fundamental tool of a mastering engineer.
—Knowledge and taste. Having the technical knowledge to be able address the problems heard in a mix and the taste to know whether or not to use a given technique.
—An accurate room and monitors. A good pair of monitors in a bad room can misrepresent a mix as much as a bad set of monitors in a good room.
—Both room and monitors work together to produce a listening environment that will not distort the presentation of a mix causing an engineer to potentially make bad decisions.
—A transparent processing chain. As with physicians, one credo of the mastering profession is to “do no harm”. Mastering engineers go through great effort and expense to ensure that their processing chain is as distortion and noise-free as possible.
—Everything from the type of cables to the software and hardware used is analyzed and potentially modified to reduce any ill-effects caused by the processing chain.
—Processing which provides additional “color”. What would seem like a contradiction to a transparent chain is the addition of hardware or software which actually adds distortion in order to enhance a track.
This includes both new and vintage hardware which adds tube distortion, transformer or tape saturation, along with software based modeling algorithms. The intent of these effects is to add warmth, thickness and depth to mixes that would otherwise sound thin or too “digital.”
9. Should you choose an engineer based on their “style”?
Ten different mastering engineers working in the same room with the same equipment will create ten totally different masters, each sounding great on their own. If you ask those same engineers to go back and reproduce any given master, you are likely to get ten almost identical masters back.
While each individual mastering engineer has his own style, it is important that he is able to separate himself from his style when needed. An engineer should never let his personal taste interfere with the goal of the artist he is working with. Again, this is where communication with the client is a crucial element.
A good mastering engineer should be well versed in a variety of different categories of music. In general, there is no reason why an engineer known for creating great country albums cannot produce a great rock album.
While an engineer’s work should be able to transcend musical genres, if a mastering engineer has a certain style that is appealing to you as the artist, you should consider working with him.
It’s important that both the engineer and the artist can communicate in a way that is complimentary to both individuals.
10. Which is more important, a technical background or musical one?
A mastering engineer should be well versed both technically and musically. The craft of the engineer is to be able to know good music and know how to make that music sound better.
Still, while a technical background is extremely important in the mastering world, that background should not interfere with the aesthetics.
Likewise, any personal feelings an engineer has about the stylistic choices of the music he is mastering should ultimately be discussed with the musician. It is because of this that an engineer’s musical background should not hinder his craft.
Given a technical background, some mastering engineers are capable of making modifications to equipment to create a more transparent sound, or provide color according to their taste and needs.
Having a musical background, particularly in the area of pitch, allows an engineer to identify frequency issues relating to musical notes and can speak directly to the musician about these issues in their terms.
An engineer should make sure that he strays away from favoring either background. While most engineers come from one or the other, their craft is in combining the two.
A mastering engineer should remain as objective as possible while still providing necessary feedback and insight from both a musical and technological perspective.
Tom Volpicelli is the president and founder of The Mastering House and has an extensive list of mastering and mixing credits to his name.
Tech Tip Of The Day: Recording Acoustic Guitar Low Frequencies
Q: I’ve been recording some acoustic guitar lately and have been having some difficulty. The problem is that I just don’t seem to be getting the low-end reproduction I was hoping for.
I know I can probably boost this later with EQ, but is there something I can do? Maybe a different mic technique?
A: There’s no doubt about it, recording an acoustic guitar can be tricky. There are several ways to bring out more of the bottom end fullness you’re looking for.
One solution — assuming your guitar has a pickup — is to record the pickup output via a DI at the same time as you record the microphones. When it’s time for mixdown, filter the pickup signal with a low-pass filter, so that all that is left is the low-end.
Shape this further with EQ, perhaps add a dash of compression, and blend to taste with the miked-up guitar tracks. Go lightly; you’ll find that it doesn’t take much of this acoustic guitar “subwoofer” track to fill out the bottom end. This trick is especially useful with detuned, alternate-tuned, and drop-tuned acoustics.
However, if you use this method you should double-check that you aren’t creating phase issues by mixing the direct pickup signal in with the miked signals. In many cases the DI will arrive slightly before the mic signals; use your ears to check whether this is affecting the mix or not. If so, slide the DI track slightly in time until the “phasiness” clears up.
As you mention, changing the microphone is another possibility. You’re best bet in that instance is to compare the frequency responses between the prospective microphones. Those which offer a more of a low-end boost are likely what you’re looking form in this instance.
For more tech tips go to Sweetwater.com
Wednesday, January 08, 2014
Producer/Engineer Noah Shain Records Breakout Artist Dead Sara On API 1608
Producer Noah Shain recently purchased an analog API 1608 small-format console to record breakout artist Dead Sara in a Malibu mansion for a forthcoming release on Epic Records.
The tracks will form the follow-up to the band’s 2012 eponymous self-release. Shain, who is best known for his work with Sonny Moore (Skrillex) and Atreyu, purchased the API 1608 because it was, in his words, “the only console available with a small enough footprint and a big enough sound.”
After recording their previous album with Shain at Sonic Ranch in Texas, the band members had their heart set on taking the laid back experience they had there to the next level.
“They didn’t want to work in a studio at all,” Shain said. “They just wanted to work in a house. I said, ‘OK, but if we’re going to work in a house, we’re going to do it right. I’m not bringing my laptop over to your buddy’s apartment.’ In the end, we rented an 8,000 square-foot Malibu mansion for two months. All told, we paid about as much going that route as we would have paid going to any of the world-class L.A. studios that I tried to talk them into originally.”
As a self-described API-fan who had been considering a 1608 to complement his hybrid, bus-intensive mix style, Shain recognized the sonic and logistic advantages of having a 1608 for the two-month remote date with Dead Sara.
“I like discrete transistor electronics,” he said. “That’s saying something, because Shain’s workflow is unique. Like Michael Brauer, Shain separately processes stems according to their role in the music.
“He uses eight passive busses and saturates signal on the other end with boutique gain make-up amps. The 1608 is the perfect heart for that kind of system.
“It’s small enough to make it easy, and I can still focus a lot on the computer for a truly hybrid workflow.”
With the remote recording date with Dead Sara finished, Shain has moved the new 1608 into his three-room studio located inside Bedrock, a 120-room rehearsal space in the hip Echo Park neighborhood of L.A.
“Bedrock is a great place with a great community of musicians,” he said. “A lot of notable bands rehearse there. My door is open and a lot of people are going to see the new API 1608.”
Posted by Julie Clark on 01/08 at 12:32 PM
Tuesday, January 07, 2014
Church Sound: Four Vital Production Tips To Propel Your Audio To The Next Level
Vital…Propel…Next Level…Can four production tips actually make THAT MUCH of a difference? Yes, they can!
The sad part is a good number of people aren’t using these tips and their sound is suffering.
Answer this question; when does your mixing work begin? Before you answer, I’ll give you three choices:; once you enter the sound booth, once you enter the sanctuary, or once you get the song list?
The problem is there are many who want to learn but there aren’t that many good teachers. That’s where this list of four vital tips comes into play. These are the simple things that should be done, could easily be done, but many times aren’t being done.
Let’s change that.
1. Mic Instruments The Right Way. I’m occasionally pulled into a church to listen to their music mix and make recommendations. Before the event starts, I check out how the instruments are miked. The wrong mic setup will have a hugely negative impact on their mix. In many cases, mixing tweaks can’t compensate for the poor mic setup.
Poor mic setups can be categorized in two forms; too far and too close. Mics that are located too far from the instrument will pick up a lot of stage noise and won’t pick up enough of the instrument. For example, a kick drum mic located too far away from the drum head would give you a dull kick drum sound and a bunch of stage noise.
Mics that are too close to the instrument can produce a distorted signal or a poor sound. For example, if an instrument microphone was set up with an acoustic piano and the microphone is placed too close to the piano strings. In this case, instead of capturing the full sound of the piano, the resulting sound is dominated by the frequencies produced by a handful of strings.
Instruments should be miked so you hear the best representative sound of the instrument and the least amount of stage noise. It’s the live environment so sound isolation isn’t possible but you do have the ability to get really close.
Oh, and make sure you’re using the right microphone.
2. Learn To Set The Channel Gain. Second to microphone location is gain setting. And gain setting is the second place where I see people make mistakes. The problem is it’s assumed the GAIN (a.k.a. TRIM) knob is a volume control and from there, it’s easy to mess things up. (Hey, I’m not judging…I used to think the same thing myself.)
How do you know if your gain settings are whacked? Do you hear a lot of hiss in a channel even when the musician is playing? Do you have feedback issues all the time? Are your fader controls normally down near the bottom of the fader slot? If you answered yes to any of these, chances are you have gain issues.
The GAIN controls the level of audio signal coming into the mixing board. Along with the audio signal, there is the presence of electrical line noise that’s part of any audio system. When the GAIN control is set too low, you hear this noise in the channel. When the GAIN level is set too high, you experience problems like audio feedback.
Each channel’s gain should be set so you have the best audio signal-to-noise ratio (S/N ratio). This means you hear a strong signal and little-to-no electrical noise.
3. Don’t Treat Musicians Equally. There is a time and a place for bias, and this is one of them.
I remember it like it happened yesterday. I watched the sound guy during the sound check and I couldn’t believe my eyes. The band took the stage, he set all of the channel volumes at the same audible volume level and then he stopped. There was no mixing or volume balancing. It was all singers and instruments coming out of the main speakers at the same volume.
In my complete guide to church audio production, I go into detail on volume balancing and there is even an audio file where I mix instruments all at the same level and then compare it to a properly volume balanced mix. The difference is dramatic. Don’t treat musicians equally!
The problem, I believe, is that what you hear and what you think you hear are two different things. For example, if the band is doing a final song that should sound big with full-on instruments and everyone singing, then you might think you should bump up all of the channel volumes.
But as soon as you do that, your whole mix falls apart because the bass is stepping on the electric guitar that’s stepping on the acoustic guitar and all the instruments are stepping on the vocals. Taking this scenario as an example, it would be better to boost of house volume so the overall balance of instruments and vocals stays the same. I digress.
How do you learn where musicians should “sit in the mix?” I’ve said this before but I can’t stress it enough; analyze professional recordings of the song. So whatever Chris Tomlin song your worship band is playing next week, get a copy of the original and listen to it over and over.
Listen to one instrument through the whole song. Is it “upfront” in the mix? Is it more of a supporting instrument in the background? Imagine all of the musicians on the stage and their location on the stage is based on where you hear them in the mix. That’s the best way to learn where musicians should “sit in the mix.”
A quick tip on evaluating your volume balance: test channel volumes by muting them.
4. Push Your Pride Aside. Whenever I find significant problems with a music mix, it’s because the sound tech messed up in one or more of the above three areas. So what about this last one?
Maybe it’s a guy thing. Maybe it’s a geek thing. Maybe it’s just me, but I doubt it. If something is wrong, I want to figure out why. If I’m learning something new, I want to figure out the how’s and why’s and where’s and all of that stuff. While I applaud anyone who desires to learn, I will applaud even more for the person who asks for help.
The last point comes down to this; no matter how long you’ve been mixing, no matter how young or old you are, there will always be something you can learn from another audio tech. And one of the best ways I’ve found of learning is by creating my mix, during band practice, and then asking another tech to show me how they would change my mix to make it better.
The Take Away
I called these four tips vital because they involve the foundation for all mixing work. You must get the first three right before starting your hands-on mixing.
As for the fourth, on being foundational…you have to have the right mindset on mixing and learning and desiring to create the best sound for your congregation. I’ve seen what happens when pride gets in the way, and it’s not pretty.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
Thursday, December 19, 2013
Addressing The Myths Of Wireless System Transmitter Power
One of the topics that I’ve seen poorly understood, and even deliberately used to mislead people, is the issue of wireless microphone transmitter power and the effects said power has on system performance.
Let’s start with the basics: all things being equal, more transmitter power = more range for the system, but not in a linear way.
In broad terms, when discussing analog wireless systems, the receiver wants to see a signal from a transmitter (carrier) that is at least about 4-6 dB greater than the noise floor before it can make use of that signal - this is known as the receiver CNR or Carrier to Noise Ratio. The better the receiver front end, the lower this number can be.
And we all remember the inverse square law, right? In other words, every time you double the distance between transmitter and receiver (in free space), you cut the amount of RF power received by four times. Thus distance is a greater factor in losing your signal than the level of power from the transmitter.
Another way to look at it is that you need to quadruple the RF power in order to have the same reception at only double the distance. This is why in the real world, twice the power at the transmitter yields only about a 20 percent increase in range.
But the other problem is that things are never equal. There are many other factors affecting the potential range of the system, from the type of surfaces nearby (or lack thereof), weather conditions, how many people are around, and of course; potential interference from other sources. Antenna system design and positioning can also significantly affect range.
In other words, transmitter power is one factor, and relatively speaking, can be a minor one. This is not to say that more power doesn’t help, because it very well may. Background noise is nearly always present, and has lately been on the increase due to DTV and other sources such as digital data services, so greater transmitter power can help to “cut through” and provide a solid signal with no dropouts.
In some cases, however, it can also be advantageous to use the lowest power you can manage with your wireless transmitters. For one thing, there is less battery draw, which may be an important factor depending upon the type of application.
Another drawback of increasing transmitter power is the resulting increase in the level of inter-modulation (IM) products. Normally, when a larger-scale wireless system is set up, careful frequency coordination is done to minimize the impact of these IM products, and usually, a certain amount of them can be ignored because they’re so low in level.
If the level of these products increases, then so does the potential for interference and reduced range.
What causes these IM products to be created in the first place? Any time multiple RF signals (such as from wireless microphones, TV transmissions, etc.) are combined in a non-linear device (any active device such as a transmitter output stage, receiver front end, antenna combiner, etc.), these signals can interact and create new signals which, although lower in level, can act just like additional transmitters.
Unfortunately, a typical multi-channel wireless setup generates thousands and even likely millions of these IM products! So keeping a handle on the level of these unwanted signals is important.
If there’s need to coordinate a large number of channels in a fairly small geographic area, transmitter power should be carefully considered, and the antenna system should be designed to match.
This particular subject has been a matter of debate in some circles. The IM problem is usually at its worst when a lot of transmitters are physically close to each other (such as in a theater situation).
Some manufacturers, such as Lectrosonics, have classically oriented their wireless mic systems towards high power transmission and solve the problem with their product designs by using an “isolator” to prevent the mixing of signals in the transmitter.
Although this does nothing to mitigate these signals from mixing in the receivers or receiver antenna systems, generally simpler, even passive antenna systems can be used effectively, along with receivers incorporating a robust front end.
Other manufacturers have maintained that “low power is better” and have oriented their system designs around this concept, with more complex antenna systems and highly selective receivers.
Another factor is that most real-world situations involve products from a variety of manufacturers (for example, IEMs from one manufacturer, handhelds from another, and belt-packs from yet another), and it is important to understand how all of the system components (transmitters, antenna systems, splitters, receivers, etc.) will react to the different power levels of the various transmitters.
As you can see, this issue is not always a simple matter of “more is better,” nor for that matter, is it always “more is worse.” Manufacturers of wireless equipment have addressed this issue in different ways, and one of the best is to offer variable power settings on transmitters, which has been done by Shure, Sennheiser and Lectrosonics, among others, in the past few years.
Mike Wireless is the “nom de plume” of a long-time RF geek devoted to better entertainment wireless system practices the world over.
Friday, December 13, 2013
The Polls Are Open! Vote In The Fifth Annual Readers Choice Awards
The first four editions of RCA racked up more than 100,000 total votes
The fifth annual Readers Choice Awards—where you can vote on your favorite sound reinforcement products—has just launched on ProSoundWeb. (Go here to vote.)
Readers Choice is unique for a number of reasons, chief among them (and as the name says), all voting is the exclusive domain of the readers of PSW.
The first four editions of RCA racked up more than 100,000 total votes. The races in most categories are close and competitive, owing to the overall strength of every product entered combined with the distinct yet varied preferences of the pro audio industry’s largest online community.
Some of the races are so tight, in fact, that they’re decided in the final days of the contest, separated by just few votes. So every vote really does matter.
The polls are open, and voting is simple - go HERE to get started.
Thursday, December 12, 2013
Blackbird Studio Chooses API 1608 For Blackbird Academy
API 1608 analog console installed at Blackbird Academy in Nashville.
An API 1608 analog console will serve aspiring sound engineers at the Blackbird Academy recording school in Nashville, Tennessee.
The school is based at the prestigious Blackbird Studio and will draw on their history, expertise, and industry connections to offer three educational curricula, totaling over 720 hours of horizon-expanding coursework.
Class sizes will be small and will feature extensive hands-on lab time, including time in two of Blackbird’s eight recording studios, now reserved exclusively for educational use. The classroom space was created by interior designers Robert and Cortney Novogratz of the HGTV television show, Home by Novogratz.
“We chose the API 1608 for a number of reasons,” said John McBride, who is co-owner and co-operator of Blackbird Studio with his wife, country superstar Martina McBride. “The 1608 is easy to use, it has a great sound, and it’s an excellent tool to teach students signal flow.
“We already have a couple of 1608s at Blackbird, and we’re very happy with them. Moving forward with the Academy, we wouldn’t have it any other way!”
Mark Rubel and Kevin Becka are co-directing Blackbird Academy, which began with a three-day summer camp designed for high school students. The Academy’s debut Studio Engineering Program kicked off on September 30, 2013 and will run for about 24 weeks.
Posted by Julie Clark on 12/12 at 01:25 PM
Wednesday, December 04, 2013
Simple Capture: Console Capabilities For Basic Live Recording
Recording a live event is pretty much the norm today, particularly with the emergence of digital technology making it a relatively simple endeavor.
Recording a live event is pretty much the norm today, particularly with the emergence of digital technology making it a relatively simple endeavor.
Basic show recordings (often in the form of a 2-track capture taken off the main mix bus that’s commonly called a “board tape”) are a handy learning tool, allowing engineers to evaluate their mixes and musicians to critique their performances.
While recordings taken from the main outputs can be quite good from a quality standpoint, there are sometimes issues. For example, instruments that are very loud in the room sometimes aren’t as loud in the PA, and since the recording is a copy of what was in the PA, the recording might not be as balanced as desired. Further, instruments that are “EQ’d” for the PA might not sound their best on the recording.
Another handy aspect are multi-track recordings used for virtual sound checks. These are pre-EQ and pre-FX recordings of each instrument and vocal that can then be played back through a system, fostering tuning and optimization without needing the band onstage.
In the corporate gig world, recordings of meetings and events may be required by law (as in the case of a shareholders meeting) or as a matter of company policy. More often than not, recordings are done so that co-workers not at the meeting have access to the material. Many corporate audio providers also make a safety recording of the event just in case it’s needed later by the client, or more likely, the video company.
While mixing consoles offer, at the very least, a few outputs that can be used to connect to a hardware recording device, today’s analog-digital hybrid and digital consoles offer a host of ways to integrate recording easily into any setup. Hybrid designs (analog units with digital features like onboard effects and computer interfaces) may offer a USB or FireWire connection to link to a computer. These consoles can do double duty, acting as a live mixer at a show while simultaneously sending audio to recording software in the computer.
The MG166CX-USB mixer from Yamaha Corporation of America is a USB-enabled hybrid. It includes built-in digital SPX effects processing and can connect via USB to a computer running the included CUBASE AI4 multi-track recording software.
Another example is the VeniceU from Midas that provides 6 aux buses, 4 audio subgroups and a 7 x 2 matrix that can all route to USB, allowing the user to easily set up recording feeds.
Record directly to iPad with Mackie DL Series mixers.
In the all-digital realm, the PreSonus StudioLive 24.4.2 console is outfitted with a FireWire port and can record each channel pre or post signal processing to a DAW. While not every digital console has an onboard FireWire port, some, like the CADAC CDC Four, are available with optional 32 x 32 FireWire interface that fosters connection to a DAW.
Many digital consoles now offer direct 2-track recording to a USB stick or an externally connected hard drive, making it easy to create a board tape or safety recording. The Behringer X32 not only ships with a 32 x 32 channel audio interface for FireWire and USB 2.0 that allow connection to a DAW, but it also offers uncompressed 2-track recording to a thumb drive via a USB port.
Don’t have a thumb drive handy? With the Mackie DL806 and DL1608 iPad-based mixers, you don’t even need one—2 tracks can be directly recorded to a docked iPad.
Another way to make a basic recording is to set up a mix either from aux buses or a matrix out. This can be a better option because it provides the ability to have a separate mix tailored specifically for the recording.
Many digital consoles offer AES and SPDIF digital 2-channel outputs that allow easy interfacing to stand-alone hardware recording devices while keeping the signal in the digital domain.
For example, with 6 stereo mixes and 4 stereo matrix buses available (plus 8 mono auxes), it’s a straightforward matter to set up a 2-track mix for recording on Soundcraft Si Series digital consoles, routed to the AES output for easy interfacing of stand-alone hardware recording devices. Another example is the Allen & Heath GLD 80, which offers 2-track recording to USB that is available from a variety of sources or the mix master, as well as a SPDIF digital output.
If you need more than two tracks but don’t have a multi-track recorder or DAW available, the Eclipe GT from Innovason offers a solution in the form of the M.A.R.S. option. It provides a built-in 64-track recorder that records directly onto a hard disk plugged into the back of the console.
Many digital consoles offer the ability to connect to a digital network. Some have their own proprietary network and others use open standard protocols. MADI protocol, which offers up to 64 channels, is a popular format used to transport audio to recording devices as well as in snake systems. Many digital systems offer MADI snakes and stand-alone recording devices like the Blackbox BBR64-MADI recorder from JoeCo that is designed to record and play back up to 64 channels of MADI data.
Plenty of I/O capability on Yamaha Commercial Audio CL Series consoles foster recording, from simple to more complex.
While the V-Mixer Series from Roland Systems Group uses proprietary REAC protocol for interconnection of mixers and stage boxes, there’s also the S-MADI REAC Bridge, a bi-directional signal converter between REAC and MADI that allows integration of recorders and other consoles into the network. In fact, you don’t even have to go to MADI to record on the REAC network because Roland also offers the R-1000, a stand-alone 48-track recorder/player designed to operate on REAC.
And, for an easy way to get MADI into and out of a computer, the UB MADI interface from DiGiCo can be used with any computer with a USB 2.0 port.
Talking The Talk
Another popular network format is Dante (from Audinate), and many consoles use it as a transport network and/or to provide interface options. Yamaha digital consoles are a good example of both.
The newest consoles from Yamaha Commercial Audio Systems, the CL Series, provide Dante connections as standard, and all of the company’s digital consoles can be outfitted with option cards that enable them to connect to a Dante network. (By the way, CL consoles offer 2-track recording by simply plugging a USB memory stick into the USB connector on the front panel of the console. No other equipment is required. Playback from the USB memory is possible as well in MP3, AAC, and WMA formats.)
Dante can also “talk” directly to many computer-based DAW programs as well as a stand-alone recorder like the JoeCo BBR64-DANTE. The new Avid S3L console uses the Ethernet AVB standard (by the way, Dante is AVB standard compliant) and can link directly to a computer running ProTools software or any other DAW, making connection for multi-track recording quick and easy.
What I’ve presented here is just the tip of the iceberg when it comes to recording options available with consoles primarily designed for live applications, and is intended to serve as a starting point to your own research. Suffice to say that we have a multitude of options available, and it’s only getting better on a regular basis.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.