Drawmer products are built to the highest standards and so it's not unusual to find our older products that are still in use even though they were originally sold thirty years ago. Here you will find information and operator's manuals for products that we no longer manufacture.
The CPA-50+ Stereo Power Amp is the perfect partner for any passive speaker, and especially in the studio with a set of passive cube monitors. The latest incarnation of the CPA-50+ has had an upgrade over it's predecessor, providing enhanced specifications that improve the audio quality and accuracy, giving it a cleaner and more articulate sound with higher clarity and detail across the frequency range, and also includes the benefit of a universal power supply.
Sitting between your monitor controller/pre-amp and a pair of passive speakers, the high performance Class D amplifier delivers 50Wrms per channel into 4 ohms (25Wrms per channel into 8ohms) and offers features including a universal power supply, thermal, overload and power up/power down protection and fault indication. With just a 6.5” square footprint, the CPA-50+ blends seamlessly with a pair of Yamaha NS10 or Auratone C5 monitor types, for example, and is small and light enough to fit into a bag to take anywhere.
In addition to working in stereo the CPA-50+ can also function as a 120Wrms into 6ohms Monoblock (set via internal jumpers), which allows the system to operate in bi-amped or bridged modes.
- 50+50Wrms into 4 Ohms.
- 25+25Wrms into 8 Ohms.
- 1x120W into 6 Ohms bridged mode.
- Use as a Stereo amplifier or as a Monoblock in Bridged/Bi-Amp Modes (via internal jumpers).
- Balanced Neutrik XLR input.
- Efficient, low heat design with Low Power Consumption. Consumes only 6W of power when idling and <0.5W when in standby mode.
- Improved specification: 115dB Dynamic Range; SNR = 101dB (1W 8 Ohms); 0.05% THD+N @ 4 Ohms; Crosstalk >80dB.
- Thermal, voltage overload and current overload protection. Standby, fault and bridge indicators.
- Improved universal mains power supply.
- Rugged steel chassis. Desktop unit that can be stacked.
- Compact and Portable.
- Dimensions: L 165mm (184mm including connectors) x W 165mm x H 51mm.
Do you hear what we hear? Our most versatile product, the HQ preamp and D-A converter straddles both the pro-audio and hi-fi markets, functioning in the pro audio market as a monitor controller, as well as serving as a source preamp in high-end hi-fi systems. The monitor preamp is possibly the most important piece of equipment for both audio professionals and Hi-Fi enthusiasts alike. It is the last piece of equipment before the speakers, so it requires zero coloration, precise balance, low noise, wide bandwidth and a wide variety of input sources.
There are three components to the accurate evaluation of audio - engineers pay a great deal of attention to both the choice of microphone and speakers. However the third component, the monitor pre-amp and digital-to-analogue conversion used to reference this material, is crucial and often overlooked. The Drawmer HQ is specifically designed to do that job - and to a highly accurate, precision-engineered standard. Likewise there are three aesthetically and functionally complimentary products that can be used independently or linked to provide multiple channels and remote operation. The three products are the HQ - stand-alone, with all controls on the front panel; the HQ-r - a desktop control surface for both the HQ and HQ-b; and the HQ-b - has no controls on the front panel but is operated by either the HQ or HQ-r. In this way several modular options are available to suit your specific requirements and setup, for example, in a studio several HQ-b’s can be mounted in a rack but controlled from the desktop using the HQ-r.
Designed to act as a central hub in a studio, mastering room, reference library or any high-end listening environment, Drawmer’s HQ facilitates fast switching between sources and outputs in multi-speaker setups. Analogue inputs accept audio on balanced XLRs and unbalanced phonos. In addition, separate stereo RIAA-corrected phono inputs are provided, so that turntables may be directly connected for restoration work or reference without the need for a separate phono pre-amp. The HQ also permits the connection of digital sources at up to 192kHz in AES format (via XLRs), TOSlink (on an RCA jack) or AES3id (on a BNC connector), and a standard USB ‘B’-type connector allows audio to be streamed at lower rates from a laptop. For the hi-fi enthusiast, he rack ears are removable, while for the rackmounting professional, the feet can be removed.
Each input source, whether in the analogue or digital domain, may be given its own input gain settings for optimum level-matching of the connected devices. Audio from mixing desks, turntables, digital audio workstations and DVD, Blu-ray or CD players may all be united within the HQ and sent to the balanced and unbalanced analogue outputs simultaneously. Digital sources at their original sample rate and bit depth can also be routed to a dedicated S/PDIF output.
So what you hear is what you get - precisely.
- SEAMLESS RELAY VOLUME CONTROL (SRVC™) - ACCURATE CHANNEL BALANCE TO 0.05dB
- STATE OF THE ART DAC (-100dB THD & NOISE AND –114dB DnR)
- ULTRA LOW CROSSTALK (>100dB @20Hz TO 20kHz)
- 4 ANALOGUE AND 5 DIGITAL SOURCES (WITH DAC TO 192KHZ/24BIT), WITH INDIVIDUAL GAIN SETTING. IT CAN SIMULTANEOUSLY CONNECT TO APPLE MAC, LAPTOP, MIXING CONSOLE, DAB FM TUNER, IPOD DOCK, MEDIA PLAYER, AUDIO WORKSTATION AND PHONO TURNTABLE PLUS A WHOLE HOST OF OTHER GEAR.
- INTELLIGENT SOURCE SELECT ALLOWS A-B COMPARISONS
- DIGITAL INPUTS TO 192kHz 24bit WITH JITTER REDUCTION
- RIAA INPUT FOR ACCURATE REPLAY OF VINYL
- SWITCHING FOR THREE DIFFERENT SPEAKER CONFIGURATIONS: SPEAKERS A, SPEAKERS B OR SPEAKERS A+B
- LINKING OPERATION FOR MULTI-CHANNEL SYSTEMS
- DUAL HEADPHONE OUTPUTS
- NORMAL OUTPUT LEVELS UP TO +18dBu BALANCED
- HOT OUTPUT LEVELS UP TO +28dBu BALANCED
- INTERNAL LINEAR POWER SUPPLY WITH MULTI-STAGE REGULATION
The A2D2 gives you high-quality A/D conversion for any professional recording application. Microphones are analog, speakers are analog, storage is mostly a digital affair, and signal processing can reside in either the analog or digital realm, depending on whether the processing is software or hardware based. At some point in the signal path, you've got to get it to the digital world, and a high quality A/D converter is required - one which faithfully reproduces analog signals in the digital domain. The A2D2's Grade 1 AES-standard internal low-jitter clock generator and Burr Brown analog input stages ensure top-quality sound from the front end all the way through the subsequent conversion and processing. This means your signal is as pure as possible throughout the recording, editing, and mixing stages.
The A2D2 is a stereo A/D converter giving simultaneous dual stereo outputs at different selectable sample rates from 44.1kHz to 192kHz. This makes it possible to have a main output sample rates of up to 192kHz with another running at 44.1kHz to serve as a low resolution copy. Automatic dither generation maintains signal purity and a linear power supply minimizes interference with the internal clocks. Three word clock outputs allow you to use the A2D2 as a master clock generator.
- Stereo A/D converter.
- Simultaneous dual stereo outputs at different selectable sample rates from 44.1kHz to 192kHz.
- Possible to have a main output at 192kHz with another at 44.1kHz as a low resolution copy.
- Accurate 24 segment peak reading LED bar meters shows incoming signal level from -50dBfs to 0dBfs plus separate overload LEDs.
- Dual input configuration allows either a fully variable input level from -2dBu to +28dBu via front panel rotary controls or 24 turn precision presets for a fully calibrated input.
- Each digital output has selectable word length of 16 or 24 bits, with automatic dither generation.
- The internal low jitter clock generator is Grade 1 AES standard.
- External clock input.
- 3 word clock outputs to allow the A2D2 to act as master clock generator.
- Burr Brown analogue input stages.
- Linear power supply to minimise interference with the internal clocks.
The MX40 Pro Punch Gate is a high performance, simple to use 1U four channel noise gate that offers the same extraordinary sonic character, ingenuity, and high technical specification that have made Drawmer the world standard in gating technology. The MX40 Pro is the first low cost, truly professional gate to offer the features and performance typical of much more expensive noise gates.
Drawing on our many years experience in professional audio we have developed a series of lower-cost Drawmer dynamic signal processors, the MX Pro range. Aimed at the musician, project studio and live performer, who is determined not to compromise on quality, but have to work to a tighter budget. By using circuit elements drawn from existing top-of-the-line Drawmer products and by simplifying the control systems where possible, the MX Pro range is no less sophisticated than our other units, despite the price tag. As a result the inexperienced user gains the benefits of Drawmer quality and performance, combined with lower cost and simplicity of operation.
The MX40 Pro features proprietary 'Peak Punch' circuitry which adds unique dynamic enhancement to the leading edge of percussive material, adding true 'punch' to the performance. The MX40 Pro also includes Drawmer's Frequency Conscious System (first introduced with the DS201 Dual Gate), which 'tunes' the gate to react only to a specific frequency band, dramatically improving gate operation. The MX40 Pro packs four channels of high power Drawmer gating into a single rack space, with controls and indicators designed to make set up faster and easier.
The Drawmer Six-Pack represents the industry’s first dedicated multi-dynamics processor for surround mixing and continues Drawmer’s tradition of audio ‘workhorse’ products which will continue to perform time and time again - even in a rapidly changing audio delivery environment.
The Drawmer Six-Pack is a single 3U unit that will provide dynamics control in any format from mono to 5.1 surround. With six independent expander/gates, six independent auto or manual compressors, six ‘zero overshoot’ limiters and an LFE channel plus the ability to link channels individually, the Drawmer Six-Pack makes surround dynamics easy. In the studio, any analogue audio source from mono to stereo to full surround can be handled in one unit. In live sound, outboard applications such as drum processing or vocals can be dealt with in one easy to use unit, with both auto and manual operation. Alternatively the unit can be re-configured at the push of a button to provide dynamic control and speaker protection without image-shift for the fast developing surround sound speaker rigs employed in the club dance scene.
- PEAK LIMITER - Programme Adaptive, variable Threshold, Zero Overshoot limiter variable from 0dB to +16dB. An LED display of limiter activity.
- COMPRESSOR - Soft-Knee compressor with variable Threshold (-40dB to +20dB), Release (0.05 to 5 seconds), Ratio (1.2:1 to infinity), Attack (0.5 to 100mS) and Gain make-up (±20dB).
- Switchable auto/manual Attack and Release with LED status display.
- EXPANDER/GATE - with variable Threshold control (-70dB to +20dB) and variable Release time from 0.05 to 1 second. Dual LED gate status display is provided.
- METERING - Eight LED bargraph meters display gain reduction, input/output levels and single LEDs display Master/Link status and Bypass.
- BYPASS - Each channel features switchable Bypass with LED status. A relay hard wire bypass system allows the unit to still pass audio when the power is off.
Many musicians and recordists appreciate the convenience of working with PC soundcards and other computer based packages but remain disappointed with the poor quality of the integral converters and processing. The Tube Stations' signal path and associated circuitry is specifically designed to capture the magic of tubes and maximise the signal level at the point of analogue to digital conversion - thereby committing the widest possible dynamic range into the digital domain.
The Tube Station 2 (TS2) is a dual mono or stereo linkable soft-knee compressor. Each channel is equipped with a variable Tube Drive control which can be switched out of the signal path for instant A/B comparison. The TS2 offers digital outputs on SPDIF or AES/EBU at up to 24 bit resolution and 96K sample rate on the optional DC1 module. Simultaneous analogue outputs are available on balanced XLR.
- 2 channel Tube soft-knee compressor.
- Single 'more compress' control.
- Variable Tube Drive.
- Side chain access.
- Fixed threshold limiter.
- 8 segment gain reduction - input/output bargraphs.
- Dual mono or stereo modes.
- Variable attack and release controls.
- Analogue output level (does not affect digital out).
- Balanced XLR +4dB, 1/4 inch Jack -10dB inputs outputs.
The ‘Signature Series’ S3 Stereo Three Band Tube Compressor incorporates the very latest in Ivor Drawmer designs and the aim from the very beginning was to create a ‘no technical compromise’ circuit using only the highest grade components.
The S3 forms the basis of a ‘Signature Series’ and offers previously unattainable control and tonality over each of the three bands - gain control at each stage provides precise spectral balancing. The signal path consists of high performance input/output transformers, passive components and 10 x tubes (8 x ECC83 and 2 x 12BH7) configured as a fully balanced Class A design. Because the LDRs (Light Dependent Resistors) in the compressors are temperature sensitive the S3 houses an ‘electronic oven’ which provides and sustains the optimum LDR operating temperature - thereby maintaining calibration accuracy and improving performance. A front panel LED indicates temperature status. Large scale VU meters can be switched to ‘Peak’ mode to show transient information. Two further VU meter re-scale modes are available (+10dB and +20dB) to accurately display the unit’s ability to output levels of up to +30dBm. The master output section includes controls for both Gain and stereo balance to compensate for source material with a Left/Right imbalance or disproportionate processing.
- FULLY BALANCED SIGNAL PATH CLASS A DESIGN.
- ISOLATION TRANSFORMERS IN AND OUT.
- 20 x ACTIVE TUBE STAGES.
- HIGH POWER ‘PUSH/PULL’ OUTPUT STAGE DELIVERING UP TO +30dBm.
- VARIABLE BAND SPLIT POINTS.
- FAST REACTING ‘OVEN’ CONTROLLED LDRS TO MAINTAIN CALIBRATION.
- ACCURACY SWITCHABLE ‘PEAK’ OR ‘VU’ METERING TO DISPLAY TRANSIENTS.
- SWITCHABLE +10dB OR +20dB METER RE-SCALE MODES.
- ‘AIR’ MODE FOR HIGH BAND.
- ‘BIG’ MODE FOR LOW BAND.
- SWITCHABLE MUTE AND BYPASS ON EACH BAND.
- INDIVIDUAL GAIN REDUCTION METERING ON ALL BANDS.
The S2 Signature Series Dual Channel Tube Compressor offers an ‘all tube - no technical compromise’ circuit using only the highest grade components. The S2 features a host of creative processing possibilities never before found in an all analogue dynamics package.
V-BIG- Retains bass frequencies and minimizes undesirable ‘pumping’ by rolling off the detection signal at 75, 125 or 250Hz (user switchable). A fully variable level control allows for the desired amount of V-BIG processing and an in/out switch provides the option to remove from the signal path for A/B comparison.
V-AIR- A dynamic high frequency enhancer to keep compressed audio sounding fresh and bright with continuously variable frequency control (500Hz to 20kHz) and variable level to control the amount of V-AIR enhancement.
DRY- Mixes user defined amount of ‘uncompressed’ signal with the compressed signal to create ‘parallel compression effect’ without the need for external mixing devices.
- 2 CHANNEL SOFT-KNEE TUBE COMPRESSOR.
- FULLY BALANCED INTERNAL SIGNAL PATH.
- CLASS A DESIGN.
- 10 (count them) VUCUUM TUBES PROVIDING 20 x ACTIVE TUBE STAGES.
- ISOLATION TRANSFORMERS IN AND OUT.
- ‘ALLTUBE’ CIRCUIT DESIGN.
- VARIABLE ATTACK AND RELEASE WITH OPTIONAL ‘PROGRAMME DEPENDENT’ AUTO RELEASE OPERATION.
- DUAL MONO OR STEREO LINK OPERATION.
- SWITCHABLE ‘PEAK’ OR ‘VU’ METERING TO DISPLAY TRANSIENTS.
- SWITCHABLE +10dB OR +20dB METER RE-SCALE MODES.
- 8 SEGMENT GAIN REDUCTION METERING.
- ‘V-BIG’ MODE FOR RETAINING LOW FREQUENCIES.
- ‘V-AIR’ MODE FOR ENHANCING HIGH FREQUENCIES.
- ‘DRY’ MIX MODE FOR ‘PARALLEL COMPRESSION EFFECT’.
- BALANCED XLR INPUTS/OUTPUTS.
The DSL424 provides the engineer with a 1U four channel comprehensive toolbox capable of solving the most complex dynamic problems. Incorporated in the 1U package are two industry standard frequency conscious noise gates and two soft/hard knee compressors with variable threshold limiting. The channels may be front panel configured as four individual stand alone processors, a stereo linked pair of comp/limiters with a stereo linked pair of gates. Alternatively, any combination of processing can be achieved by rear panel patching.
The DSL424 combines the gates of the DS404 with the compressor/limiters of the DL441 to provide a versatile 1U problem solver invaluable in applications where rack space is limited.
Recognizing that there are many general applications which require equipment that is simple to operate, Drawmer have designed the noise gate stages of the DSL424 utilizing ‘Programme Adaptive’ circuitry. This makes the DSL424 ideally suited for use over a wide range of input signals, ranging from drums and other percussive instruments through to vocals, pianos and even complete mixes.
The provision of variable Low-Pass and High-Pass Filters allows ‘without compromise’ frequency selective gating with a KEY LISTEN facility enabling monitoring of the filter setting.
Each gate channel can be operated as HARD - offering ultra-fast response time, stable triggering and a specialized release contour which is ideally suited to percussive material or SOFT - a versatile Expander capable of handling vocals and sub mixes.
Variable High/Low Pass Filters for frequency triggering.
Switchable range with LED status.
Selectable ‘Hard’ (drums etc.) or ‘Soft’ (vocals etc.).
1/4 inch jack Key Inputs.
Key Listen facility.
The compressor section features a highly developed, auto attack/release system which adapts itself to the programme material being processed and is equally effective on individual sounds or complex mixes.
To maximise flexibility, each channel of the DSL424 is switchable between traditional ratio (hard knee) and soft-knee operation. Traditionally, soft-knee compressors are preferable for unobtrusive level control or for the control of finished mixes, whereas ratio type compressors are generally considered more successful in creative applications or where large amounts of gain reduction are required. By offering a choice of both modes, the DSL424 is capable of outstanding results in a very wide range of studio and live sound situations. The two channels may be operated independently or linked for true stereo operation.
Also included in each channel of the DSL424 is a peak limiter which allows the user to set an absolute output signal level that will not be exceeded. If the peak limiter threshold is exceeded for more than a few milliseconds, additional gain reduction will be applied to reduce the overall signal level to within accepted limits without distortion. Once the peak has passed, the system gain will return to normal after a period of about one second. This facility is extremely valuable both in live sound applications, for driver protection, and in digital recording where an absolute maximum recording level exists. Furthermore, when deliberately overdriven, it can be used creatively to produce level “pumping” effects which can be useful on electric guitar or rock vocal sounds.
Switchable Hard/Soft knee compression with variable threshold, ratio and output gain.
Variable ‘zero overshoot’ transparent limiter.
High resolution bargraph metering of gain reduction and input/output level.
Balanced +4dB XLR in/out.
The benefits of multi-band processing, with the ability to apply different parameters to different sections of the audio bandwidth have been apparent to sound engineers for quite some time. However, ‘locking in’ your audio to any one specific multi-band unit can vastly reduce the creative options. The desired combination of processing to deal with low frequency energy, the vocal mid range and high frequency detail and enhancement is not necessarily available from any one manufacturer.
The new Drawmer THREE-SUM opens up a whole new set of options allowing the engineer to split the audio into two or three bands and apply his or her own sonic signature to each part of the audio bandwidth. The THREE-SUM employs a high quality signal path culminating in a variable threshold, brick wall limiter section with bypass facility. To ensure transparency the limiter design is ‘two stage’, applying different processing to the H.F. content of the material. In applications where the dynamics of the material need to be retained to create an open sound, the limiter is essential for catching peaks.
The THREE-SUM has been designed for use in high-end mastering and general recording applications, alternatively, it can transform a project studio, equipped only with ‘single band’ processors, into a serious multi-band facility.
Continuously Variable Band Splitting.
Switchable Mute & Bypass On Each Band.
Variable Threshold, Two Stage Brick Wall Limiter Section.
Two Or Three Band Operation.
VU Metering With +10dB Re-Scale Mode For Hot Output Levels.
Variable Input & Output Levels.
Individual Band Or Multi-Band Processing.
High Quality, Minimal Signal Path With Balanced XLR Inputs/Outputs.
In today's studio where increasing numbers of digital devices need to be interconnected, maintaining digital signal integrity can be a serious problem. The traditional professional solution has been to use an expensive master clock that drives all the digital equipment in the studio, but in many different types of studios there are often pieces of equipment that don't have wordclock inputs such as consumer CD players, MD recorders, DAT machines and budget computer soundcards.
Drawmer's unique solution to this dilemma is the M-Clock, an affordable yet extremely highly specified, multiple output master clock generator that also incorporates four further channels of sample-rate-conversion, all locked to the same master clock. There are also alternative sync outputs for those pieces of equipment without wordclock connections that can be sync'ed via their digital audio inputs using a 'black' signal carrying no audio information.
The advantages of combining wordclock with multiple channels of 'clock-locked' sample rate conversion are significant and often under-appreciated. With M-Clock, all your 'pro' and consumer digital equipment can run in the same system, locked to the same clock, with no worries about finding a suitable sync source. You'll find you're no longer having to switch equipment between internal and external sync as you repatch - and the M-Clock's ultra-precision, low-jitter clock helps maintain optimum stereo imaging, lower noise and lower distortion. If you have multiple pieces of digital equipment in your studio, whether professional or a mixture of professional and consumer, M-Clock is probably the most effective upgrade you can make for anything like the price.
A High-stability Master Clock and Dual Sample Rate Converter. The M-Clock Plus is a high stability master clock generator offering clock rates from 44.1 to 192kHz, coupled to two sample rate converters, which allow material to be simultaneously re-sampled and syncronised to the selected high precision clock.
In addition to the internal clocks, M-Clock Plus can sync to either external word clock or clocks from AES/EBU signals. Precision clock frequency measurement and display indicates the exact frequency of the selected clock, whether internal or external, to an accuracy of 2ppm.
Ten word clock outputs
The M-Clock Plus has a total of ten word clock outputs, so it's ready to handle even the most complex systems. Eight outputs are located at the rear and two are on the front panel for convenient access.
Various format and sample rate conversion possibilities
M-Clock Plus has an external word clock input and an AES11 input which retrieves the clock from an AES audio signal. Plus, each sample rate converter has selectable AES, SPDIF or TOSLINK inputs, with simultaneous AES, SPDIF and TOSLINK outputs for many format and sample rate conversion options.
Convenient display information
The M-Clock Plus' precision clock frequency measurement and 16 character blue LCD display indicates the exact frequency of the selected clock up to 768 kHz, whether internal or external, to an accuracy of 2ppm or 2 microseconds. Display modes include ppm, ±ppm & % pull up/down for video users.
The one-rackspace TS1 features Class-A Drawmer preamps and a stereo soft-knee Drawmer tube compressor, coupled with a 24-bit/96k digital converter. Usable either as a stand-alone stereo-tube compressor or as a dedicated voice or instrument channel preamp/compressor, the TS1 features a HF contour control and variable highpass filter on the inputs, and a variable Tube Drive control for adding tube saturation. Additional features include balanced analog I/Os, phantom power, an effects insert point and compressor sidechain. A optional digital-output module adds S/PDIF or AES/EBU at up to 24-bit resolution and 96kHz sample rate.
The MasterFlow DC2476 is a state of the art digital mastering processor drawing on Drawmer's considerable experience in analogue signal processing and offering un-parallelled power and flexibility at sample rates from 32kHz to 96kHz.
All the key functions are modelled on classic analogue signal processors with the added benefits which digital processing allows.
Signal processing functions are as follows:-
Input Level adjustment and stereo balance.
Full band Bootstrap Compressor/Expander.
5 band parametric Analogue modelled Equaliser with selectable Bell/Shelf filters for Bass and Treble bands.
3 Band Expander.
3 Band Bootstrap Compressor.
3 Band limiter and 3 Band Stereo Width.
3 Band Tube Saturation.
3 band output level control. Auto Fade with selectable fade shape and Rounding to 16,18,20,24 bit using 4 selectable dither shapes.
Although a wide range of manual control is available, all Expanders and Compressors are programme adaptive, taking account of the dynamics of the signal being processed. Since the object of a Mastering processor is to push the signal as close to full scale as possible and increase the overall loudness and spectral detail, any overload, anywhere in the system must be prevented. This is accomplished by a Gain Management system which monitors signals throughout processing and reducing levels when overloads could occur.
Most 24 bit A - D converters are hard pushed to give 19 bit resolution - the other five bits are essentially marketing bits. However, by using out proprietary Multiple Converter Technology, we have produced a 24 bit A - D conversion package with a massive 130dB dynamic range and extremely low distortion. With built-in sample rate conversion, the DC2496 can output signals at 44.1, 48, 88.2 or 96KHz sample rates while implementing bit-rate reduction using one of the best sounding dithering strategies available anywhere. A high quality D - A convertor is included for accurate monitoring of the digital signal and the facility is included to enable true 24 bit/96KHz audio to be recorded in stereo using six tracks of an ADAT, DA88 (or other multitrack recorder using a compatible interface). These recordings may then be played back via the DC2496 to reconstitute the original 24 bit, 96KHz signal. An additional 16/20 bit 44.1K/48K In/Out is provided to record a simultaneous back-up.
Proprietary ‘Multiple Converter Technology' giving a true 130dB dynamic range at 24 bit, 96KHz.
AES/EBU, SPDIF, TDIF and ADAT interfacing.
96KHz/48KHz D-A for accurate monitoring.
44.1, 48, 88.2 or 96KHz sample rate plus Ext Clock.
Simultaneous 16/20 bit 44.1K/48K back-up output.
Digital Gain with four mode limiter for max level.
Sample rate and format conversion including ADAT to TDIF and vice versa.
Enables 24 bit/96KHz stereo recording on ADAT and Tascam TDIF machines via proprietary format.
The 1962 incorporates perfectly matched pairs of pre-amps and A/D convertors and the inclusion of a switchable 'zero overshoot' transparent limiter enables the full dynamic range to be utilised, without any fear of digital overload.
The 1962 can be supplied analogue only - with the digital section available as a retro 'slot-in' module.
Although the 1962 employs a minimal signal path philosophy, it is also packed with innovative processing features which may be switched into the signal path for creative applications.
ZERO OVERSHOOT LIMITER ensures that even the most difficult signal is kept under control. Digital overload is impossible.
VARIABLE Hi and Low PASS FILTERS, for the removal of rumble and any unwanted high frequencies.
FINE TUNE EQUALISATION. 3 band of EQ, Low Mid and High, with ±10dBs of cut or boost.
DUAL DYNAMIC ENHANCE. Two variable controls to add warmth and sparkle to the signal with no increase in the overall signal level. Individual indicators to show activity.
VARIABLE TUBE DRIVE allows the amount of 'Tube Sound' to be adjusted to suit the material being processed.
MIX output section incorporating PAN pots for stereo width control and stereo output level, with metering. Either the individual channels or the mixed stereo outputs can be routed to the Digital convertors. A further 3 signals can be patched into the mix section via the rear panel.
Huge LED bar meters with 24 LEDs plus additional Peak LED indicator. Separate Channel and Mix output metering.
Each processing section has individual BYPASS switches for simple A/B comparisons.
Separate analogue output level controls on both Channel and Stereo Mix outputs.
INPUTS: Each channel incorporates MIC with +48v phantom power and Phase Reverse; Balanced LINE input and AUX (instrument) input, assignable from the front panel. Rear panel send / return INSERT jacks are offered.
STEREO LINKING for accurate tracking of the stereo image when processing stereo material.
Digital resolution is selectable 24, 20, 18 and 16 bits. Switchable sample rates of 48kHz and 44.1kHz plus a superior locking and tracking EXTERNAL SYNC for other master clocks.
DITHER or NOISE SHAPING. Fifteen selectable algorithms are offered to apply to the digital output, with formats and levels to match all types of programme material.
Industry standard AES/EBU, SP-DIF and TDIF outputs.
Intelligent digital Word sync input that monitors the quality and condition of incoming clock.
Fully CE compliant.
The 1962 is ideally suited as a front end for Digital Workstations, 'Direct to Digital' classical recordings, stereo mastering, two-channel sound reinforcement mixing and many more broadcast and studio applications.
The Drawmer DF330 is a single ended noise reduction system which does not require any previous encoding. Instead, it uses a dynamic filter which senses the highest frequency of incoming program material and opens a filter to that frequency. It has been designed to dramatically reduce noise, regardless of source, in a wide variety of recording/broadcast situations. An example of this is when old recordings are remastered for CD. Some other applications would include: Analogue to digital mastering, reducing tape hiss and noise resulting from high gain and high frequency boost on mixing consoles. Restoring degraded or poorly recorded material for tape duplication and resolving problems with noisy synthesizers, guitar amplifiers, effects units etc.
The main features of the Drawmer DF330 include:
Excellent visual displays are provided. The input level is displayed with a 3 colour 'traffic light' style LED meter. The filter cut-off frequency display has nine LED segments and the expander gain reduction has a five segment LED display. Further to this there are LED's to show bypass, stereo link status and power input
A 50Hz 'rumble' filter with a slope of 18dB per octave is available independently on both channels
In addition to a full system bypass, the expander and filter can be switched out of the audio path separately, enabling easy comparisons to be made. under all conditions
A comprehensive expander with automatic attack time, variable release and switchable ratio and attenuation gives up to 40dB reduction in overall noise during silent passages
Stereo linking allows both channels to track identically. This prevents image shifting with stereo programme, and phase shifts when both channels are mixed into mono
The dynamic filter has two modes of operation. In the manual mode, the threshold is adjusted by the user to give optimum results. In the Auto mode the unit is capable of tracking the signal level thereby enabling the filter to continue to operate successfully during fade ups and downs. This makes manual adjustment unnecessary
Simultaneous -10dBu unbalanced 1/4 inch jack and +4dBu balanced XLR inputs and outputs allows the unit to be used as a level converting interface between the two operating levels, e.g. The output from a cassette machine at -10dBu could give a balanced +4dBu output and -10dBu output
Discontinued in 1996 but still much sought after.
The Drawmer M500 provides everything you could ask from the most powerful digitally controlled dynamics in the world and more. Its ability to combine up to seven 'effects' simultaneously is unsurpassed.
A different combination of effects and parameters can be assigned for each of its two channels. Utilising two high specification VCAs to provide a minimum signal path, the Drawmer M500 eliminates the disadvantages of limited bandwidth, noise build up, and time delay associated with multiple signal processing.
Ease of Operation.
To ensure that the engineer can exploit the full potential of the Drawmer M500 in the minimum of set up time Drawmer have included special features to allow 'user friendly' operation.
128 memory patches to provide instant access to 'everday' processing tasks.
All parameters of operation can be adjusted by a single rotary encoder.
'Colour coded' keypad for function selection.
Backlit 'supertwist' LCD for easy readability.
FREQUENCY SELECTIVE NOISE GATE- Offers a powerful array of parameters including peak attack, trigger, pre-delay, re-trigger mask and 'Dynamic Envelope Transfer' for fully creative signal shaping.
DE-ESSER- Full band, single band, two band and two band complex.
COMPRESSOR-Switchable Hard/Soft knee compression, selectable Auto threshold, Auto attack/release and Auto gain make up.
EXPANDER-Threshold, ratio, hold, release, and range.
LIMITER-Threshold, attack, hold and auto/manual release.
AUTO PAN-Adjustable parameters and 8 pre-set pan waves triggerable from any source including MIDI.
AUTO-FADER- Fade up time, fade down time, range and trigger source (up to 99 seconds fade time). Keypad or MIDI triggerable.
78 factory pre-set memory patches.
50 user memory patches allow the engineer to edit, title and store a library of specialised programmes.
User modifications to a factory pre-set can be compared to the original pre-set using the EDIT/RECALL key.
Within the noise gate function 16 envelope memories are available for 'DYNAMIC ENVELOPE TRANSFER'. The dynamic envelope (amplitude) of any sound can be simply recorded and imposed onto any audio signal passing through the gate.
Comprehensive 'REAL TIME' MIDI control.
High and low pass side chain filters.
Intelligence to prevent 'parameter abuse'.
All threshold parameters can be displayed simultaneously.
Balanced XLR connections.
Many studios start as a small concern, but quickly expand into serious, productive enviroments. Often, this means selling some of the budget equipment, in order to replace it with equipment of the necessary quality to match the upgraded studio. The Drawmer LX20 is designed to serve the needs regardless of the size and quality of the studio.
Soft knee. The ratio is increased automatically as the compression increases. Only one control is required to adjust compression.
Expander to eliminate low level noise. Release time linked to Compressor Release.
Selectable system level -10dBu or +4 dBu.
Side chain access.
Variable Attack and Release.
Eight element GR meter. Three element signal level meter.
The Drawmer E101 was a four stage, single channel passive equaliser, followed by a stage of makeup gain to make up for the insertion loss of passive circuitry. Traditional Inductor/ Capacitor/ Resistor circuits are employed in the filter design and a very low noise, solid state amplifier is used to provide the necessary gain. This in our view, gave the best of both worlds; the smoothness of purely passive filtering combined with the low noise, low distortion amplification which can only be accomplished using recent technology. The format of the Drawmer E101 offered a high degree of control at the extreme of the audio spectrum, where precise correction or subtlle enhancement is often necessary.
Because the Drawmer E101 used a passive equaliser circuit, seperate filter sections were provided for cut and boost. This had the advantage of allowing the user to simultaneously add cut and boost at different frequencies. With all the cut and boost controls set to flat (fully anti-clockwise), the equaliser had unity gain.
There are four equaliser sections, two for the bass end of the audio spectrum and two for the treble. Frequancy selection is by means of stepped rotary switches while the cut and boost controls are continuously variable.
BYPASS switches the EQ and make-up gain amplifier out of circuit.
BASS CUT The Bass Cut section employs a shelving Hi-pass filter which may be switched to operate at : 20Hz, 30Hz, 60Hz, or 100Hz. Up to 15dB of cut may be applied and this section is useful for attenuating frequencies below the useful range of the programme being processed or for reducing overall boominess.
BASS BOOST Switched to operate at : 20Hz, 30Hz, 60Hz, or 100Hz., the bass boost section offers a choice of Hi-pass shelving or peaking characteristics. In 'Peak' mode the 'Q' of the filter is 2.5, and the maximum boost is 16dB. In 'Shelf' mode the maximum boost is 13.5dB.
TREBLE BOOST the Treble Boost section comprises a variable 'Q' peaking filter operatring at frequencies of 3KHz, 4KHz, 5KHz, 10KHz, and 16KHz. The bandwith control allows adjustment of 'Q' over the range 0.8 to 2.1. Up to 12 dB of boost may be applied at the minimum 'Q' setting, and up to 18dB of boost are available at the maximum'Q' setting.
TREBLE CUT This section has a shelving Lo-pass characteristic and may be switched to operate at 5KHz, 10 KHZ or 20KHz. A maximum of 18dB of cut may be applied. Apart from its tonal applications, this section is useful for attenuating frequencies above the natural range of the programme being equalised which will also serve to attenuate any high frequency noise present in the source programme.
DL231 - Dual Expander/Compressor
The Drawmer DL221 was a dual channel hard knee compressor with limiter. The Drawmer DL231 was a balanced version of the Drawmer DL221 with an expander.
The Expander had variable threshold from +10dB to -40dB. Three switched ratio settings of 1:2, 1:5 and 1:20. Three switched attenuation levels of -10dB, -20dB and -40dB. Expander metering was the 'traffic light' style as used in the DS201 noise gate.
The compressor section had threshold down to -24dB, ratio of 1:1 to 20:1, attack of 50mS to 5 seconds, release of 50mS to 5 seconds and gain of -20dB to +20dB. A ten element universal meter could be switched to show VU or GR.
The limiter section was selectable by a toggle switch. It had a fixed level of +6dB (+10dBU). A single red light showed limiting action.
The T102 interface is an 'add on' unit which can increase the already considerable versatility of the Drawmer DS201 Dual Gate, but may also be used with other types of equipment. Examples include: delay lines, auto-panners, reverb units, drum synthesisers etc.
It is a well known fact that some MIDI keyboards, sequencers and drum synthesisers suffer from the disadvantage that some of their MIDI data is incompatible with other MIDI devices. The MIDMAN was designed to overcome this problem by becoming the central control unit which controlled all attached devices.
The first Drawmer product, very rare and highly prized. Prospective purchasers should check that the unit is in good working order. Restoration or repair could prove very expensive.
The Drawmer DMT 1080 contains two independant delay lines. A multi section delay having a maximum delay time of 80mS, has taps at 10mS, 30mS, 50mS and 80 mS which are fed to either channel 1 or channel 2 by the touch button selectors. The same tap on both channels produces a mono output whilst different taps produce stereo images. Chorus on either channel produces two delays, each of which is differant from the main 10,30,50,80mS delays and the two chorus delays on the other channel; so chorus on both channelsgives a stereo image.
The output from an auxilliary 10mS delay is fed via the 'Phase' level control to both channels giving a mono image. A clean feed signal is connected in a similar manner via the 'Feedthrough' level control.
Delay time is variable over a 30:1 ratio either manually ('Manual Delay') or automatically using the internal low frequency oscillator which runs at 0.05Hz to 15Hz.
Control Voltages for each delay line are in anti-phasei.e. when the main delay is at maximum, the 'Phase' delay is at minimum and vice versa. This means that true phasing can occur at several points in the sweep (from min to max delay) as each main delay tap in turn cancels the signal from the auxilliary delay.
Outputs from the main delay taps are of fixed level. They are mixed with the variable 'Phase' and 'Feedthrough' signals before the output level controls. + or - signals from channel 1 can be routed back via the feedback control to the delay line inputs. +feedback produces resonant peaks at short delays whilst - feedback gives a notch filter type of response.