Mainley SLAM User Manual

OWNER'S MANUAL
MANLEY
SLAM!
Stereo Limiter And Micpre
MANLEY
LABORATORIES, INC.
MANLEY LABORATORIES, INC.
13880 MAGNOLIA AVE.
CHINO, CA. 91710 TEL: (909) 627-4256 FAX: (909) 628-2482
http://www.manleylabs.com
EveAnna's email: emanley @ manleylabs.com
Tech Support: service @ manleylabs.com
REV. 11-22-2004
CONTENTS
SECTION PAGE
INTRODUCTION 3, 4
POWER SUPPLY 5
BACK PANEL & CONNECTING 6, 7, 8, 9
FRONT PANEL 10, 11
METERING 12, 13, 14
LIMITERS, HINTS, ETC. 15, 16, 17, 18
DIGITAL RAMBLINGS 19
THE GUTS, INSTALLING THE A/D/A, ALIGNMENTS 21, 22, 23, 24,25
TROUBLESHOOTING 26, 27
MAINS CONNECTIONS 28
SPECIFICATIONS 29,30
ADDENDUM FOR SLAM! MASTERING 32,34
WARRANTY 35
WARRANTY REGISTRATION 36
APPENDIX 1 - EXAMPLE SETTINGS 37
APPENDIX 1 - TEMPLATE FOR STORING SETTINGS 38,39
2
INTRODUCTION
THANK YOU!...
for choosing the Manley SLAM!. This unit combines Mic and Instrument Preamps, 4 limiters, comprehensive metering and is ready for or already has the digital converter option. As one might expect, the basic operations are fairly simple and instructions may not be needed - but - the SLAM! has a lot of advanced features, and we strongly recommend reading through the manual. There are a lot of tricks and features that are not so obvious.
In truth, the SLAM! started with the idea of an updated Electro-Optical Limiter and the original working name was ELOP II. That didn't last long. First we developed a fast FET limiter, decided that fast LED metering was approriate, then added a mic pre, decided that this box would make the ideal "analog insert in a digital world", then added almost every request and suggestion the customers had given us over the years. And somewhere during all this, decided each little part had to be right, and much was going to be quite new and elaborate. In the end, ELOP II was not at all descriptive and after a 'name this box' contest on our website it became the SLAM!.
We can start right at the basic tube circuits. These designs are unlike any others we know of, including previous Manley circuits, so this is not a box with an old mic pre combined with an old Opto Limiter and a borrowed FET limiter with a conventional digital converter tossed in the salad. This is all new. The tube circuit is a hybrid FET/ tube design first used in the Manley Steelhead phono preamp and provides the advantages of both technologies. You get the low noise of FETs, the headroom of tubes, the gain of both and lower distortion than either typically, and a new texture in your tool kit.
This is the beginning of the story and continues through the product and the manual. Some manuals seem to imply that if you use 'this box' then you are an instant mastering engineer or top producer with all the tools they use. This strange manual is filled with warnings, caution flags, grumblings about some aspects of digital and has extended quotations from other other manufacturers. Our intention is to help the user, supply a bit of under-reported info, and give equal time to both what might help provide the sound you've been looking for and what might be considered questionable or dangerous to your music. The SLAM!, like other powerful processors, can be great or horrible depending on how it is used or abused and if something here helps avoid disasters, then we have happy customers.
GENERAL NOTES
LOCATION & VENTILATION
The Manley SLAM! must be installed in a stable location with ample ventilation. It is recommended, if this unit is rack mounted, that you allow enough clearance on the top of the unit such that a constant flow of air can move through the ventilation holes. Airflow is primarily through the bottom panel vents and out through the top.
You should also not mount the SLAM! where there is likely to be strong magnetic fields such as directly over or under power amplifiers or large power consuming devices. The other gear's fuse values tend to give a hint of whether it draws major power and is likely to create a bigger magnetic field. Magnetic fields might cause a hum in the SLAM! and occasionally you may need to experiment with placement in the rack to eliminate the hum. In most situations it should be quiet and trouble free.
We also suggest that you get familiar with the back panel switches and jacks before it gets mounted in a rack. If you have the digital option, experiment with the filter settings, dither, etc to find your favorite settings, then rack it.
WATER & MOISTURE
As with any electrical equipment, this equipment should not be used near water or moisture. Beer is OK though.
SERVICING
The user should not attempt to service this unit beyond that described in the owner's manual. Refer all servicing to your dealer or Manley Laboratories. The factory technicians are available for questions by phone (909) 627-4256 or by email at <service@manleylabs.com>. Fill in your warranty card! Check the manual -
Your question is probably anticipated and answered within these pages...... RTFM
3
The Swiss Army Knife
First Things First
The SLAM! is an unusual product that doesn't quite fit into a simple catagory. We get questions like "Why have a mic-pre on a limiter?", "Why have a DAC on a mic-pre?" and "Why so many input and output jacks?" and "Why no hard-wire bypass on this mastering
processor?". And the only answer is "It's not just a ....., it does a lot
more". It isn't a channel strip - no EQ, besides being stereo. It isn't just another front-end for the workstation. It isn't just a mastering processor. Maybe the SLAM! is a new catagory.
The SLAM! is intended to be the reference analog I/O (input/output) for a digital studio - the first choice analog insert for digital with strengths as an input device, output device and killer go-louder box. Sometimes digital gets cold and sterile, and people reach for tube processors for particular vintage colors, the 'warmth factor', the ballz, the thing that plug-ins or digital processors are not quite doing for 'em. So the SLAM! has an outrageous D/A and A/D option which sets it up as 'THE' Insert, and can be used with any other analog (or digital) gear to process tracks already on hard disk and then return them as pristene or mangled as desired. This requires great converters and analog circuits that can be super clean or dirty or in-between. The DAC can be pristene high-end solid state, tube, or driven hard for a wide range of colors. The ADC uses a transformer (iron) for its front end (and no chips) and is intended to be a 'warm' converter, because most everybody has the other kind.
The SLAM! is a an outboard limiter and a new low-noise high gain tube mic-pre, and a mastering processor, and a DI, and possibly the best converter in your rack. As a mic pre it offers about 70 dB of gain and a new circuit, unlike any previous Manley PreAmp. The gain stages are based on a circuit developed by Mitch Margolis for the SteelHead phono pre-amp. Mitch also designed the VIPRE. The SLAM! can be used as a mastering processor (not a multi-band limiter), a processor that real mastering engineers use to create loudness without messing up the mix. As a DI or Instument Input it offers 2 impedances 100K and 10 meg ohms, plenty of gain, limiting, and if you want to have fun use both channels with your fave EQ inserted between, and use the optional A/D straight into the workstation.
Swiss army knife is the most appropriate description. A multi­purpose, well constructed, generally useful tool. The analog to digital converters also have the Swiss connection. They are designed and built by a Swiss company called Anagram Technologies who are not well known in pro-audio but getting an enviable reputation in the high-end audiophile market. Stereophile Magazine in April 2002 (p77&79) named them the 'kings of digital filters' and they build converters for Audio Aero, Camelot Technologies, Nagra, and Manley.
What makes Anagram's converters so special? The DACs up­sample to 192K and in the process remove jitter almost completely and to the point where A/D/A worst case jitter components are well below the -144 dB measurement limit of 24 bit digital. The analog to digital converter similarly has a permanent sample rate of 192K and then down-samples to any of the common data rates like 48K. This provides both the audible benefits of 192K and the practical benefits of a 48K data rate, like relaxed requirements for giga-bits of storage space. This is all done with proprietary software running on a pair of very fast SHARC DSP chips and 40 bit floating point math.
We only have a few simple suggestions for your first few dates with the SLAM!.
1) Don't rack mount it until you are familiar with the back panel and have experimented a bit with the jacks and switches that you might use later. No problem racking it, but this way is easier at first.
2) Watch those levels. There is a lot of gain and ways to manipulate gain on the SLAM!. We have seen guys set up 30 dB of boost to a line signal, 30 dB of limiting and were not aware of how drastic those settings might be because they were unfamiliar with the box. On he LED meters, one segment = 1 dB (approximately), and if you see the LEDS go half way down, you are hitting 13 dB of limiting which is generally drastic. Most engineers prefer 6 dB or less limiting. You need to use your ears, and your eyes. Common mistake.
a) Unity gain for line inputs is near 12:00 for the INPUT and OUTPUT controls. Begin with the ELOP and FET thresholds fully clockwise (5 oclock). A good starting point. b) To set INPUT levels start with the VU on INPUT and the VU attenuator at the "0dB" especially as you become familiar with the SLAM!. You have to be aware that practically all the knobs and switches affect level and gain and that you want to start off on the right foot, so get the INPUT set first. Then set Thresholds and Output level. Most early confusion has been due to level settings. c) The LED PEAK METER (audio mode) is most useful to view when setting up the limiters and comparing how much louder it can get while hitting the same peak level. Compare your original peak level in Bypass to the level possible with limiting engaged.
3) This is a limiter and limiters generally can create weird distortions especially when the gain reduction is deep and releases are fast. The SLAM! FET limiter has very fast releases so it can be dangerous. The OPTO is easier to use because the attack & release are slower which is why opto's have always been popular. Sometimes we want the ease of opto and the speed of FET, and using the FET gently to 'clean up' the overshoots of the opto is pretty easy too. With the FET limiter alone, some experimentation and critical listening is a must. Different songs and sounds seem to want different settings and one may often be surprised by the optimum setting.
4) Because the SLAM! is old-school analog, the limiters won't have the 'precision' of a digital limiter that can be easily set to hold peaks within 0.1 or 0.2 dB of clipping. If you intend to use it as a brick wall limiter before the A to D converter as a method to be safe/lazy/ clever, in an attempt to get hot levels within .2 dB of digital clipping you may be creating the worst case scenario for an analog box. It is difficult to set the SLAM! up to do that. It can be pretty good IF you take the time to carefully set the controls. Foolproof and easy - no, but if you want 'easy', then the safest way is to accept -2 to -5 dB DFS (23+ bits), and use a digital limiter like an L1 or L2 for the last few dBs. The combination provides the best of both worlds. Another approach is to try the "CLIP" setting plus the OPTO which is a bit easier and may or may not be as audible. It might not be worth being obsessed with hitting -.1 dB DFS and focus on the sound instead.
5) Once you have found your favorite back panel settings, feel free to rack mount the SLAM!. Yes, you can leave Phantom on all the time. Old consoles didn't have phantom switches and it was always on - no problem. Most guys stick to one sample rate, 24 bits, and one choice of filters.
4
MANLEY SLAM!
POWER SUPPLY UNIT
electric shock, do not expose this unit
refer all servicing to qualified personnel only
WARNING:
to reduce the risk of fire or
to rain or moisture.
CAUTION:
risk of electric shock.
do not open, high voltage.
line voltage
switch
replace fuse
with same
type and rating
FUSE
ON
N9512423
5
4
3
2
POWER
1
1) POWER MULTI-PIN: 16 PIN AMP connector that screws into the matching socket on the back of the SLAM!. This should be connected first. Rotate the whole connector until it mates with the socket, then just a turn or so on the outer ring clockwise will complete the mating. Force will NOT be needed. The cable is 6 feet long and keeping the supply 6­12" away from other gear reduces the possibility of induced hum, though this supply won't radiate much. The supply may get reasonably warm, and this is an intentional trade-off to keep those magnetic fields minimal. NOTE: The bulk of the power supply will not turn on (including the LED) unless this connector is inserted, because the SLAM! remote controls the power for most of the Power Supply Unit.
2) IEC POWER SOCKET: Use the supplied IEC cable to connect the Power Supply Unit to wall current. This supplied cable should be the proper type for your country.
3) POWER TOGGLE: "ON" is marked. Note that BOTH this toggle has to be in the 'ON' position and the SLAM! front panel red 'POWER' button has to be pushed to turn on the SLAM!.
4) FUSE: The fuse protects you and the SLAM! in case of a catastrophe. Replace only with the same value and type. Failure to do so voids the warrantee. For 117 volts AC mains this fuse is a 2 Amp standard 1/4" SLO-BLO MDL 2. For 220 volts AC mains this fuse is a 1 Amp standard 1/4" SLO-BLO MDL 1. A blown fuse often looks 'blackened'. Several blown fuses indicates a problem needing repair. Both the Supply and SLAM! should be returned to the dealer or Manley Labs for service if the correct value fuses continue to blow.
5) VOLTAGE CHANGE-OVER SWITCH. This should be set to the proper voltage for you country by the factory. The switch is marked, and in general a good habit to verify this setting is correct before plugging any new gear in and turning it on. The carton the SLAM! arrives in will also be marked for 117 or 220.
In case you were wondering about the handle, it is to mechanically protect the switches and fuse. The 4 mounting screws allow the supply to be mounted in a rack or screwed down to something.
5
This page will have connection example diagrams
6
THE BACK PANEL
INSTRUMENT
DAC OUTPUT
UNBAL OUTPUT
FULLIN = +4dBu
HALFIN = -10 dBv
IN =100K OHM
1/2 IN =
1/4" BYPASSES
XLR O/P. XLR
DRIVES ADC
CHANNEL 2
ALANCED INPUT
B
10 MEG
TRANSFORMER
BALANCED OUTPUT
EXT LINK
(5.1 etc)
I/P OPTO
CHANNEL
2
1
SEND
SC O
RTN
OPT
SEND
FET SC
INSERT INSERT
RTN
EXT LINK
)
(5.1
SC MIX
SC MIX I/P FET
CHAS
GROUND
CIRCUIT
XLR PINOUT: TRANSFORMER BALAN PIN 1 =
SIS
PIN 2 = HOT = PIN 3 MIC = 2400 OHMS,
1/4 PHONE JACKS : BAL INSTRUMENT = UNBAL, 100K INPUT Z 
DAC OUT = BALANCED, +4dBu UNBAL OUT = +4 dBu (not -10 dBv) (INSERTING PHONE JACK DISABLES XLR O/P. XLR O/P CAN THEN BE USED FOR DIRECT ADC INPUT (SEE MANUAL
SERIALNUMBER
DESIGNED BY HUTCH
ANAGRAM TECHNOLOGIES,
JERRY GARSZVA, E. MANLEY
M. MARGOLIS, B. HERNANDEZ
CED
SHIELD = GROUND
POSITIVE PHASE
PHASE
= LOW = NEGATIVE
LINE = 50K OHMS
ANCED INPUT = 2400 / 50K OHMS
HALF OUT = 10 MEG OHM FET
)
PHONE (909) 627-4256 FAX(909) 628-2482
ELECTRIC SHOCK.
CAUTION -RISK OF
DO NOT OPE
TU
MANLEY LABORATORIES, INC
13880 MAGNOLIA AVE., CHINO, CA 91710
N. REFER SERVICIN
ONLY
UALIFIEDPERSONNEL
Q
DO NOT HOT PLUG RN OFF POWER SUPPLY
m
www.manleylabs.co
G TO
ANTOM
PH
POWER 48V
OFF
MUTE MONITORS
THEN PULL TOTOGGLE
ADC
DAC
FILTER
FILTER
40K
NORM
80
20K
20K D
NORM
44.1
ADC
SAMPLE RATE
88.2
48
DITHER
OFFOFF
96
80
20K A
AES/EBU IN RATE
WORD CLOC
SUPER CLOCK
WORD
LENGTH
ADC
K
K
CLOCK
24
20
16
K
NOISE SHAPE
WORD
AES
EBU
I
N
AES EBU OUT
IN
CHANNEL 1
ANCED INPUT
BAL
TRANSFORMER
BALANCED OUTPUT
IN =100K OHM
1/2 IN =10 MEG
1/4" BYPASSES
XLR O/P. XLR
DRIVES ADC
INSTRUM
DAC OUTPUT
BALANCED +4
UNBAL OUTPUT
FULLIN = +4dBu
HALFIN = -10 dBv
ENT
1 17
24 23 22 21 20 19 18 11 12 13 14 15 16 8 9 10 7 2 3 4 5 6
This is just a description of the various jacks and switches on the back. We suggest that you might want to not rack-mount the SLAM! until you've spent some time becoming familiar with the various patching and switches. Sorry, but the back panel is more complex than most gear and there are options galore. That's what happens when you get what everybody asks for.
For simple maps that show a few examples of how to patch the SLAM!, check out Page
1) POWER CONNECTOR. First verify the POWER SWITCH on the front panel is off (out) and Outboard Power Supply is off (toggle towards FUSE). The Outboard Power Supply has a captive 6 foot cable, with the mating plug. There is a painted white dot on the plug that should face UP and/or gently rotate it to find the "key" or where it fits. Force is definately not needed. Rotate the ring on the front of the connector CLOCKWISE about 1 turn, which locks the connector in place.
The SLAM! uses a trickle of power to remote control the power of the main supply. This means the Outboard Power Supply can not be turned on without it being connected to the SLAM!. This is a safety feature. Also, with that exception and the +&-6 volt supply used for tube filaments (heaters) all other pins are protected from passing current or a charge stored on power supply capacitors. In other words, its pretty safe, and that you can't get a shock, and can't power it up hot but still better to have both power switches off to connect this plug.
2) CHANNEL 1 INPUT). This is a Neutrik Combo jack that accepts XLRs, 1/4" mono phone plugs, 1/4" balanced phone plugs. This is both the LINE INPUT and the MIC INPUT. Phantom Power (17) is not advised except for some MICs, and in particular FET condensor mics and some rare exceptions that require it. If Phantom is ON, turn it off before patching into this jack (or any other mic patching) or at least turn the monitors, headphones, etc down because there will be a loud speaker killing POP. Contrary to urban myth, it is highly unlikely Phantom Power will damage any mic or cause it to sound different, except during patching. It is a good idea, not to have Phantom on for LINE inputs, as it might be possible to damage something with the 48 volts if you select MIC (which can enable phantom).
3) CHANNEL 1 OUTPUT. This is a transformer floating balanced +4 dBu output (Pin 2 hot). It is equally happy feeding balanced or unbalanced inputs but for unbalanced inputs, be sure that the XLR's Pin 3 is grounded or connected to Pin 1 or the shield. There is also a transformerless unbalanced 1/4" phone jack output described below (6).
4) INSTRUMENT INPUT. This is a 1/4" mono unbalanced high Z input for guitars, basses, synths, etc. It has about 30 dB more gain than the Combo jack input and uses the mic-pre. The input impedance is 100K suitable for synths and guitar processors/amp simulators but might be a bit dull for some guitars direct. For very high Z (10 meg ohm) physically insert the jack half way. Cool trick, huh? This will sound brighter for many guitars and basses, but should have little or no effect if any electronics are between the guitar and input.
5) DAC OUTPUT. (Left) This is a direct balanced +4 dBu output of the DAC if that option is installed. We calibrate the output so that Digital Full Scale = 8.0 volts and that Digital -14 = +4 dBu. Other calibration or reference levels requires a simple internal trimmer tweak.
6) CHANNEL 1 UNBALANCED OUTPUT. This jack provides an unbalanced +4 dBu output pre-transformer. It can also provide a semi- pro or consumer -10 dBv with plug inserted half way. Most other Manley gear makes this unbalanced output jack entirely bypass the transformer and thus disables the balanced XLR output. There is an internal jumper to provide this mode, if needed (see page XX). In the SLAM!, the transformer is also used to drive the A to D converter, which we did not want to 'mysteriously lose' when they plugged into the 1/4" output. With the jumper in the alternate position (expert mode), this 1/4" jack can be used as the main output to feed an EQ, and returned to the male 'output' XLR which will feed the A to D directly and passively. We like 'expert mode' but can't ship it that way. ___________________________________________________________________________________________________________
FAQ - Why no -10 input when you provide a -10 output? There is no dedicated -10 dBu input but both the Combo jack (2) and Instrument Input (3) can be used in conjunction with the INPUT knob. Why share the same XLR for both MIC and LINE and no Phantom Switch on the front? It was originally, but we added the HP filter & DAC on that switch, and felt it was 'safer' for your speakers to have the phantom on the back (see item 17). Why no separate MIC-PRE output? 3 reasons, the mic-pre would require a good line driver (no more space for more tubes), the opto limiter works between both 'sections', and the Combo jack is used by both mic and line. In other words, it required 4 more XLRs and 2 tubes for this box and there is no room. 2 channels & 24 jacks (26) plus 6 switches already, and gotta draw the damn line somewhere.Why use bantam jacks for side-chain inserts? (see 18)
7
7) DIGITAL OPTION: Some SLAM!s will have this converter option and some will not and have just a blank panel fitted. The 'Option' is two parts, and one is this panel with jacks, switches, and its 3 circuit boards. It is intended to be something the average user can install and similar to inserting a PCI card in your computer and attaching a few ribbon cables to it. The second part is a 5"x6" board that contains the converter chips, clocks, PLL, two very high speed SHARC DSPs, and a micro controller. These boards are static sensitive and one must be grounded if handling them. Like, don't be shuffling your feet on the carpet as you go to pick it up.
8) AES EBU INPUT: The standard digital input that feeds the D to A converter (DAC). It accepts data or sample rates of 44.1K, 48K, 88.2K and 96K. The sample rate is asyncronously up-converted to 192K and jitter is removed in the process. A dedicated high speed SHARC DSP chip running proprietary code and 40 bit math is used for that. The result is near zero jitter, and audibly less 'time smear'. Note 1- The DAC outputs are the 1/4" phone jacks (5) and/or can be passed through the tubes, limiters and iron as needed via the SOURCE switch on the front panel. Note 2- We have used simple XLR to RCA adapters to use SPDIF outputs to feed this jack and it seems to work fine but there are true SPDIF to AES converter/adaptors that would be the officially recommended method.
9) WORD CLOCK INPUT: Regular BNC jack that accepts a master word clock or Super Clock. In complex workstation installations often a distributed master word clock is used to guarantee stable and accurate timing across the variety of digital gear in use. With most DACs, this usually helps improve the sound.The most common report is an improvement in imaging. The biggest reason is that it offers a better alternative than AES lines for carrying the clock component, which can be considered an analog signal, and most converters have less than perfect clock recovery or jitter removal circuits (PLLs) (being very polite here). The other reason is that converter chips have what is called 'fixed' coeffecient FIR filters that depend on highly accurate clocks and stand crystal oscillator clocks are not usually that accurate. Because the Anagram converter is almost immune to jitter, and uses 'adaptive FIR filters' and not the internal (free) chip filters, the usual audible benefits of the word clock may not apply to this converter. However, probably other components in your system like word clock and this converter retains compatibility (and convenience) by the inclusion of this input.
10) AES EBU OUTPUT: This is the A to D converter's (ADC) output. As with the DAC, the actual sample rate is 192K which is down­converted to your choice of data rate from 44.1K to 96K. A second SHARC DSP is used here.
11) DAC FILTER: In developing this converter, our research (and Bob Katz) suggested that often the biggest audible differences between converters depended on the designer's choice of filters, so we gave you the choice. The toggle switch provides 3 different filter frequencies, 20K, 40K and 80K passive analog circuits. The 20K and 80K are 3 pole 18dB per octave, and the 40K is 2 pole 12dB per octave, each based on the Manley Massive Passive filters. A good starting point is 80K in a great system and 20K is less harsh / brittle and perhaps warm.
12) ADC FILTER: Similar ideas as the DAC filters except the 40K setting has a 1dB bump or boost at 20K for 'air' and a unique feature and is otherwise the steepest of the 3 settings. Note with both 80K filters that because the maximum data rate is 96K and Mr Nyquist's theorem, the maximum true bandwidth is about 45K. If the only concern with filters was the final bandwidth, the 80K would be pointless but because of a variety of factors there does seem to be subtle audible nuances between all 3 filters even at 44.1K data rates (20K BW).
13) SAMPLE RATE: The actual sample rate is always 192K and this knob is really a "DATA RATE" control. The seven settings are:
44.1K, 48K, 88.2K, 96K, AES EBU IN RATE (locks to the DAC input AES stream), WORD CLOCK, and SUPER CLOCK, (locks to the clock rate input to the BNC connector). Why no 192K selection? There is no official standard for 192 connections and the defacto method is 2 XLRs and obviously we didn't have room on the panel, besides we provide the audible benefits of 192K without having to use that high of a data rate. The converter is 'ready' for 192 if it becomes more common and connection to 192 systems becomes more feasable.
Why no WORD CLOCK output? Experts agree that the best A/D clock is its internal crystal, and this ADC is always in that mode even if you lock to the word clock input or AES. As described above, part of the reason better word clock generators sound better than cheaper ones is due to the absolute accuracy (48K= 48,000.0000 Hz) and you are better off with a dedicated box with oven controlled crystal accuracy than the built-in clock of this converter, and if desired, the better boxes will lock to the AES output of this converter and allow proper distribution required. In other words, if you are going to do it - do it right, or don't do it. Always listen, and don't blindly trust word clocks.
14) NOISE SHAPE & 15) DITHER: The ADC samples at 24 bit resolution, but often we need 16 (or 20) bit data. Simply chopping off the 8 bits results in audible artifacts and a loss of low level resolution. Modern converters add a small (almost inaudible) amount of noise (random numbers from +1to -1) called dither which effectively smooths the data and removes artifacts (trading audible distortion for a tiny amount of noise). In other words, the signal better appoximates the original analog. Noise Shaping is used to describe two different techniques that also help. The first is a method where the dither is pushed to the near ultrasonic and most of the noise energy is focused where we are least likely to hear it so we still retain the benefit of dither but don't hear the noise. The second method uses the difference between the 24 bit data and the 16 bit data (those 8 bits) which can now be considered an error signal, and does the same thing - pushes the energy to the extreme high part of the spectrum, which also seems to increase resolution. Some companies refer to this as 'apparent resolution' and claim, for example, 19 bit resolution on 16 bit data. The biggest difference can be heard on reverb tails and the end of fades and there may be personal preferences involved so it is worth evaluating for yourself. None of this applies with 24 bit word lengths.
16) WORD LENGTH: This toggle sets the ADC and AES EBU OUT word length for 24 bit, 20 bit or 16 bit data. Most recording today seems to happen at the highest resolution of 24 bits, but mastering to CD still requires 16 bit data and some MPEG codecs prefer proper 16 bit data rather than 'raw' 24 bit. The NOISE SHAPE and DITHER described above are only active if 20 bit or 16 bit is selected.
8
17) PHANTOM POWER SWITCH: This simply turns on 48 regulated volts of phantom power that 'rides' on Pin 2&3 of the BALANCED XLR INPUT. In this case, it is also only ON when you select one of the 3 Mic modes on the SOURCE switch. However, we do have several warnings: a) Because you can and will have a typical line input often plugged into that XLR and because you can easily switch to MIC, and because there is a chance some gear is not designed with DC blocking capacitors (or they are rated for less than 48 volts) there is a chance of doing damage to line level gear by 'accident'. We don't know of this ever happening but can imagine that it is possible. b) In general, patching mics with phantom turned on is a habit to break. Mic signals are typically 1/100th of a volt, and phantom is 48volts so rather huge speaker killing pops are likely - unless monitors, headphones, etc are turned way down or off. c) For the same reason as above, running mics through patchbays, intermittant cables and corroded XLRs with phantom turned on may be extra noisy and crackley. If you need phantom, you need good solid connections. The only mics that need phantom are most FET condensor mics, and some other internally preamplified mics and a few DI boxes. We don't know of any dynamic mics or tube condensor mics that require phantom. d) Contrary to urban myth, we also don't know of any mic that can be damaged by phantom, whether it needs it or not, except a few 'modified' vintage ribbon mics that had their protective capacitors removed. Early Neve and Trident consoles applied phantom power to every mic jack and offered no switch to turn it off. It is probably also a myth that some mics sound better with phantom off, but not a myth that bad jacks and cables will sound better with it off. Use phantom power only if its needed.
18) GROUND TERMINALS: These provide separate grounds for use in some installations, with special star grounds or other grounding techniques to prevent hum. In most situations the two terminals are simply connected with a wire. The top terminal marked CHASSIS is the AC third pin MAINS ground which also connects to the chassis's, rack rails and can internally connect to XLR pin 1. The bottom terminal marked CIRCUIT is the internal audio ground, which also connects to the 1/4" jacks sleeves.
19) SIDE-CHAIN INSERTS AND LINKING. These are all regular Bantam jacks, like are used in most patch-bays. Why these? Again size, space and they offer true, no-BS inserts like a patch-bay does. Most studios have Bantam to XLR adapter cables (we used to chop long patch cables in half and solder on XLRs) and if they don't, they should. All of the outputs are impedance balanced (30 ohms), single ended +4 dBu signals. The inputs are single ended, high Z, with the ring connected to ground through a 30 ohm resistor and should be compatible with most pro gear, balanced or unbalanced.
Some engineers like to patch in an EQ into the side chain of some compressors or limiters which alters how the limiter responds. For example if the EQ is set to boost at 6K (or HP filtered at 3K), the limiter becomes more of a De-esser. Some dynamic controllers seem to be extra sensitive to low frequencies and bass, so we filter out low frequencies to prevent excessive pumping or squashing on bass heavy material. A text-book limiter would not have side-chain inserts because it is supposed to accurately limit true signal peaks. The SLAM!'s Opto Limiter has a switch that provides some side-chain low freq filtering at 100 and 200 Hz. The 200 Hz setting also boosts about 4 dB at 6K, for a bit of gentle de-essing to be the "vocal setting".
Because there are actually 4 limiters in the box, side chain inserts require 8 jacks (sends and returns). The two top jacks are sends for the Opto limiter (L&R) and they are half-normalled to the two returns below them. Some may also use these as alternative outs from the MIC PRE, but the Opto Limiter and, to a lesser degree, the FET Limiter will affect them, but it avoids the final tube stages.
The next 4 jacks are similarly used for the FET Limiter. The Send is an op-amp isolated version of the unbalanced main output. Any plug or patch cord in any of the 'returns' breaks the normal and unless there is a healthy +4 dBu signal inserted there, you won't see any limiting.
The jacks marked EXT LINK (5.1) are just intended for those lucky guys with 3 SLAM!s who need a way to link 5 or 6 limiters for surround work. Two regular Bantom patch cables are required. These jacks are parallelled, and the rings carries the Opto audio link, the tip carries the FET DC link. The LINK toggle on the front panel must be in the BOTH & EXT position and all controls on all 3 units are used. The Opto Link blends all 6, the FET link uses the moment-to-moment highest signal of the 6 channels.
The bottom two jacks are unused at present, but might be useful for mods and special versions. We had an idea to use them for a 'blend' input because some guys like to use the drum sub-mix to 'push' the 2 buss limiters, but we felt this was a bit excessive and can be done with the above side-chain inserts easily enough.
20) CHANNEL 2 BALANCED INPUT: Similar to Channel 1 described by 2) above.
21) CHANNEL 2 BALANCED OUTPUT: Similar to Channel 1 described by 3) above.
22) CHANNEL 2 INSTRUMENT INPUT: Similar to Channel 1 described by 4) above.
23) CHANNEL 2 DAC OUTPUT: Similar to Channel 1 described by 5) above.
24) CHANNEL 2 UNBALANCED OUTPUT: Similar to Channel 1 described by 6) above.
9
FRONT PANEL
DI LINE DAC
SOURCE
ELOP
LIMITER
12
6
MIC MIC 180 MIC 100Hz
INPUT
200
SC
100
HZ
FLAT
OUTPUT
20
26
STEREO LIMITER AND\U+2569MIC PRE, & MORE
2
1
0
3
5
.1
SecmS 75 50 25
10
DIST
RELEASE
VF
A T T
F
A C K
FET
M
LIMITER
12
6
26
10
.2
20
.5
1
1.5 2
20
40
20
0
STEREO LINK
DUAL MONO
BOTH & EXT
60
80
100
G R
MODE & RESET
1
2
3
20
SLAM!
LIMIT
POWER
LIMIT
PEAKLED
LED TRIM
LEFT
RIGHT
2
1
0
3
5
77
10
40
20
0
VU
1
2
80
100
60
G R
VU 0 dB
VU -6 dB
-3 dBO/P
I/P
DI .1
MIC
LINE
3
DAC
MIC 180 MIC 100Hz
SOURCE
SC HZ
ELOP
LIMITER
12
20
6
26
INPUT
200
100
FLAT
OUTPUT
HANDCRAFTED IN CHINO, CALIFORNIA
mS Sec
.2
75
.5
50
1
25
10
1.5 2
DIST
RELEASE
VF
A T T
F
A C K
FET
M
LIMITER
20
12
6
26
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 1 2 3 4 5 6 7 8
1)SOURCE: This is the Input Selector that you use to choose the input to the SLAM!. The choices are DAC (digital to analog converter
if this option is installed), LINE, DI, MIC, MIC ø (phase reverse), and MIC 100 HZ (high pass filter) which has a little graphic showing the filter. LINE selects the BALANCED LINE INPUT Combo jack (XLR or 1/4") and is intended for +4 dBu signals, but by cranking the INPUT level can be used with -10 dBv signals. DI selects the INSTRUMENT INPUT jack and routes it through the mic preamp for lots of gain if needed. MIC also uses the BALANCED LINE Combo jack and routes it through the mic pre for 60 dB of gain (and another 20 dB by cranking the OUTPUT level). MIC ø is the same except opposite and just phase reversed (the proper term is polarity reversed). MIC 100 Hz is normal polarity but the lows below 100 hertz are filtered out which is useful on many vocals and overheads, to remove pops, air conditioning rumble, etc.
2) INPUT: This is the first volume control and has about 60 dB of range for MIC and DI, and from -20 to +20 for LINE and DAC. For LINE and DAC, the normal setting will be 12:00, or straight up, but this isn't the rule or an absolute calibration. For MIC or DI, the knob might be anywhere depending on the mic, the loudness of the instrument, the distance, etc., and it might be prudent to turn the knob down to start, rather than starting at 12:00.
3) OUTPUT: This is the final volume control and is used to set the output level to tape or disk, and as the 'gain make-up' after the Opto Limiter, if you need to compare 'limit and bypass'. The FET Limiter senses the signal right at the output jack, so it acts as if it is a final limiter after the OUTPUT level. Circuit-wise the FET Limiter is directly after the Opto, and before the OUTPUT level (but doesn't act like it) and before the final tube gain stage/ line driver. The range of this knob is about -20 dB to +20 dB with unity gain near 12:00. Most of the time it will live between 12:00 and 3:00.
4) OPTO LIMITER: A simple threshold knob for the OPTO limiter. Fully clockwise (+26) is 'out' and a good place to start. As you turn this knob counter-clockwise, there is more likelyhood that limiting will happen. Some dynamics units have the threshold go one way and some the other. On the SLAM!, all of the pots, should make the signal louder when turned clockwise, (except the RELEASE which is a switch).
5) SC HZ: Side-Chain Hertz. This is a HP filter in the Opto Limiter side-chain that makes that limiter less sensitive to low frequencies. It does not affect the FET Limiter. The filter helps minimise pumping and strange volume changes. Sometimes kick drums and bass seem to 'trigger' too much limiting. The FLAT setting, bypasses the filter, 100 filters 100 Hz by 6 dB and more for frequencies below that. Similarly 200 filters 200 Hz 6 dB and more below but also boosts about 4 dB at 6 kHz for gentle and subtle de-essing and can be considered a vocal setting. Normally, 2 filters like these require one to change the threshold significantly, but these are compensated to minimise that.
6) FET LIMITER: Another simple threshold knob. One can blend any balance of Opto and FET limiters by using the two threshold controls. Each limiter has its own character and advantages, and they complement each other, so that by using both, one can get most of the advantages without the disadvantages. For example, the Opto can limit deeply, smoothly and has a high 'ratio', but is a little slow for drums, while the FET Limiter can be very fast, but not as deep. The Opto has inherant time constants but the FET can be adjusted for attack and release times. What the Opto misses, the FET should catch, depending on how you blend them and the FET ATTACK time.
7) ATTACK: This just affects the FET Limiter. With compressors, 'attack knobs' are used to set how fast the compressor responds and pulls down a signal. Traditional limiters don't have this because it compromises the 'concept' of limiting if there is any overshoot. We compromised somewhere between 'text-book limiter' and 'typical compressor', and simulated much of the 'sound' of the attack control while still providing more transient reduction than is apparent. VF is very fast (.1 mS), F is fast (1mS) and M is moderate (10mS). VF is the best if you need to prevent 'overs' and is closest to the traditional or text-book limiter. F and M tend to let more transient through but are also more punchy and may be less detrimental to drums. Expect to adjust the FET LIMIT knob a bit for similar depths of reduction when you change ATTACK (typical). There is another side-chain that grabs much of the peaks, almost inaudibly, but our ears tend to hear the side­chain that has the ATTACK and RELEASE knobs. Use your ears to determine the best setting. Instruments with fast transients like drums show the biggest differences, vocals less so, and soft flute-like sounds may not be affected except for a little threshold difference.
10
8) RELEASE: This only affects the FET Limiter. There are 11 positions numbered from 2 Seconds (slowest), to 10 milli-Seconds (fastest).
Slow releases tend to be the least audible and will be cleanest. Medium release times on the SLAM! are pretty fast for a limiter and where the most loudness increase tends to be, but if pushed too far also might be obvious with pumping or a volume rise after the 'cresendo'. This may also be near the edge of when 'modulation' starts to become audible, especially if there is a lot of bass energy in the signal. Achieving maximum loudness cleanly is not automatic and might require a bit of play between threshold(s) release time and attack because it really depends on the music. The SLAM! attempts to minimise all the negatives, pumping, modulation, loss of 'energy' that is typical for a limiter with fast attack and release times because this is where the maximum loudness lives - but - this is dancing on the edge of a dangerous cliff.
The SLAM! release time can be set up for ridiculously fast releases (10 & 25 mS) that pretty much guarantee modulation distortion with lows, which is most often undesirable but can be used as an effect and yet another paint brush. We might caution using ultra-fast release times with bass instruments, but it can be fun on rude drums and blazing solos. There is also a CLIP setting, which introduces a FET clipper that is fairly round like some low feedback tube circuits overdriven and is a bit reminiscent of speaker distortion. We wanted to provide a psycho-acoustic memory of loud, and this is one way. The CLIP is best suited to enhance a moderately distorted guitar, of fatten a synth. It is not intended to replace your Marshall, or amp simulator, but can often be used to take them a bit further.
9) STEREO LINK: A 3 position toggle. The center position disables stereo linking and is labelled DUAL MONO. All of Manley's previous limiter/compressors provide a LINK switch and both L&R controls have to be used for proper operation. Meanwhile, most other compressors just use the left side while the right side controls become useless. Enough people requested, for us to include this mode of LINKing. This is the STEREO LINK or up position. Both ways have advantages. The modern 'left-side only' is convenient, easy and can be clever especially on a plug-in. The problem is that almost all implementations mono the L&R, which means sounds that are hard right or left are 6 dB less likely to trigger limiting than sounds down the center, and anything out-of-phase won't be seen by the limiter at all. We think a proper 'mastering compressor' is supposed to react to the peak waveform of both the left and right equally, or stop the same peaks that causes the A/D to clip. This is easy in digital, but in analog it requires the user to use both sides, and that the limiters react equally based on whichever side has the loudest peak. So the SLAM! also has that mode "BOTH & EXT" or the down position. This mode is also used for the back panel linking to other SLAM!s for surround projects. For recording instruments the STEREO LINK mode is fine but for serious mastering the BOTH & EXT mode is usually best. No LINK of the CLIP functions because why have one side clip the other?
10) LED (meter): This switch controls the LED bar graph meter. In the center position is basically an PEAK display of the audio output. The upper position is basically to display GR (Gain Reduction, especially the FET Limiter). The down position is a momentary switch that RESETs the peak hold (clears the dot) and is used to select the LED meter MODE if held down for a few seconds. A full and complete description of the LED Meter is on page 12. Suffice it to say here that it does a lot.
11) LIMIT LEFT: Push it in to engage both the OPTO Limiter and the FET Limiter and the OUTPUT level control and it lights up blue. This is not a hard-wire bypass, nor can it be, on a Swiss Army Knife, that has multiple inputs and outputs, mic pre-amps, etc. A hard-wire bypass on a mastering version is a bit more likely.
12) POWER: OK, we won't do a 300 word description of a power switch this time. Push it, it lights RED, and turns on the bulk of the Outboard Power Supply, which has been on idle drawing almost zero current. (If it doesn't, remember that there is also a power switch on the power supply that has to be turned on.) The VU meters should light up, and about 30 seconds later the MUTE relay disengages (to prevent tube warm-up thumps) and audio should be available or rising gently. This box has a long warm-up time but should be stable in a minute and very nice in 15 minutes. Power-down mutes immediately. This might be a concern in a live situation, plan accordingly.
If you are not using it for 8 hours, you might as well turn it off to save power bills and tube life. There is a school of thought that suggests that the initial turn-on is the hardest on tubes, and shortens their life and to some degree that is true. From our experience, it all depends on the individual tube and some last 30 years and some 30 seconds. If you are concerned with tube life and down-time, repairs etc, buy a set of extra tubes and save yourself some panic when you least need it. Changing a tube is almost as easy as changing a light bulb and once the top cover is removed should take 20 seconds (compare that to arepair needing the 'ol soldering iron).
13) LIMIT RIGHT: Just like 11, but for Channel 2. Push to engage limiting and the OUTPUT level.
14) VU: Selects the source for the VU meters. I/P (input) shows the level directly after the INPUT level pot and is a good place to set the
MIC-PRE gain or rough out operating levels. O/P (output) shows the output level appearing on the output jacks. GR shows the OPTO Gain Reduction, but not the FET. Most Opto Limiters use a VU to display gain reduction. When the limiters are bypassed the VU drops to below
-20 which is not intended to imply extra hard limiting. The Opto can also be displayed on the LED meters, with an expected increase in speed because the Vactrols in the Opto Limiter are faster than VUs.
15) VU ATTENUATE: One can also pad the VU's down by 3 or 6 dB which is especially useful if the client is in the room and eyeing the VU needles pegged in the red. Mastering engineers need this because a final mix has a lower ratio of peak to average level, and probably lower again after mastering. The SLAM! will tend to do that too. For individual tracks, we expect the 0 or no pad to be the most common but it depends on the instrument and distorted guitars may few and moderate peaks, but some percussion has huge peaks. We suggest the 3 dB pad for most mastering, and allowing about 3-6 dB below digital full scale on the peak meters. Why? FIR filters usually need 1dB to 2 dB of headroom, MPEG usually requires 4-6 dB, the mastering engineer needs some room to work. The 6 dB VU pad is a hint that maybe this is project is 'hyper-compressed', especially if is still bending far into the red. It is nice for CD playback though.
11
METERING
The SLAM! has some very comprehensive metering. If you skip this section, you’ll be back here with the yellow highlighter pen, once you start really using the box.
There are both LED bar graphs and standard VU meters, and each can show a variety of information. Before proceeding further, we should mention that any peak meters and VU meters should look different with music and that they are intended for different purposes. VU meters are deliberately slower, and are mostly intended to represent apparent loudness much like the way our ears work. LED peak meters are most often chosen when very fast peak reading signals need to be represented. For “standard” VU meters there is a long list of specs and qualifications, from needle size, color and ballistics to meter size, color and scaling. Most importantly, VUs have been around a long time and most engineers find them easiest to interpret and most valid for analog tape. Because peak meters are fast compared to VUs, transients like drums will look louder than on VUs, and this is a good thing if we are concerned with clipping or digital recording.
The LED bar graphs are multi-color (8), multi-mode (4), and multi-purpose. The meter is controlled by a 3 position toggle switch labeled LED, with MODE & RESET (down), PEAK (middle) and GR (up). During normal operation, the switch will be in the center or up position and will control the display on the meters based on the currently selected mode. The momentary down (spring) position has different functions depending on the current operating mode and how long the switch is held down.
Holding the switch down for more than .5 seconds suspends normal operation and enters the Mode Menu, indicated by a colorful pulsing on the top 2 segments of the right meter. While in this menu, one of 4 Modes described below can be selected. The selected mode is shown by a lit segment surrounded by two others near the bottom of the right meter. You can scroll through the 4 modes by pushing the switch down again quickly (before it returns to ‘normal’). Once the desired mode is lit, you ‘confirm’ the choice by either waiting 2.5 seconds or once again pushing down the switch but longer than .5 seconds. Modes 1 and 2 are the normal display modes and normal operation resumes. Modes 3 and 4 are used to select options.
Modes
MODE 1) DUAL DISPLAY: (shows two things at once)
Peak Position (center): Displays audio as a green and amber bar from the bottom up. FET Gain reduction is simultaneously displayed as a red dot from the top down. GR Position (up): Displays FET Gain Reduction as a Green bar from the top down and OPTO Gain reduction as an Orange dot from the top down. The Peak Hold dot and the third bar color are not available in this ‘dual display’ mode
In each mode ADC clipping is also displayed. In Mode 1 PEAK the Green segments turn Red. In Mode 1 GR, the Green FET bar turns Red. In Mode 2 you can have the Green turn Red or the top 2 segments turn bright Red depending on where the LED TRIM is set.
MODE 2) SINGLE DISPLAY:
PEAK Position (center): This is a typical peak meter with a peak hold dot. The bar is divided into 3 colors, Green, Amber & Red. GR Position (up): Displays the sum of FET and OPTO Gain Reduction (or the total limiting) from the top down. Both the limiters are added together which may look a lot more drastic than it sounds and some interpretation is required. It is most useful when minimal limiting is the goal.
MODE 3) COLOR CHANGE POINTS:
You can change where the audio peak meters change color from green to amber and from amber to red. This is an unusual feature of this meter and lets you 'match' the SLAM!'s peak meter to the meters on your digital recorder or work-station. The SLAM! meters are analog and your other meters are very likely pure digital so exact segment for segment matching is unlikely. This just gives you a way to set color change points.
Select Mode 3 with the momentary toggle and the display will change so that the left meter is totally lit and the right meter is still in audio peak mode (for reference). Below the POWER button is a small hole and a trim pot lives behind it. A small screwdriver or tweaker is needed to adjust it. If the Mode Switch is in the center position, then you can adjust the point where green changes to amber. If the Mode switch is in the GR or up
position, you can change the amber to red point.
If you run out of range on the trim, exit Mode 3, tweak the trim to approximately center, then go back to Mode 3 and set the color change point where you want.
If you tweak the orange-red color change point to "over the top of the display" (max), then the ADC clip indication for Mode 2 changes. Instead of the top 2 segments turning bright Red, all the Green segments turn Red when the ADC clips at digital Full Scale.
MODE 4) PEAK HOLD MENU
This is just for the peak hold dots in Mode 2. The selection is displayed on the left LEDs. a) No peaks held (no dot) (indicated by the lower of the 3 left segments) b) Peaks held for 1 second (typical generic peak meter) (indicated by the middle of the 3 left segments) c) 'Infinite' Hold, stores the peak until you manually reset by a quick push down and release of the momentary toggle. This mode is handy when the SLAM! is behind you and you need to know after the fact, the loudest transient that went through. Unfortunately, it won't hold a ADC clip indication.
RESET
Each time a mode is changed, all settings are saved to non­volatile memory. If you find that you have 'out moded' yourself, or that the meter is looking goofy, the factory default settings can be restored as follows. With power off, press and hold the LED Meter switch down, and turn the power on. The left meter will light 2 segments red, the right green, release the switch.
Sometimes the Opto will cause a stuck segment because the opto GR is not slowed down (shared from the VU meter GR mode). Just change to Peak and back will clear the segments, or wait until a the next hot peak clears it for you.
12
Loading...
+ 27 hidden pages