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Psychoacoustic Bass Enhancement Unit with Renaissance Compressor, MaxxBass, and L2
Bass Boost for Live Sound, Broadcast, Mastering & Post Production
They say that in electronics there's no new technology - the only changes
will be in size and applications. Things will simply keep getting smaller
while doing more. This is certainly true in music (just think about what an
iPod can do), but as we design smaller playback systems with small speakers,
there are also some tradeoffs, particularly in music. Clever cabinet designs and
high-excursion drivers can get you only so far. Any audio engineer trying to extract
bass from real-world components must eventually face the laws of physics:
All things being equal, small boxes and speaker cones can't move enough air
to produce a room-shaking low end. When working with systems tiny enough to
toss in a shoulder bag, the challenge is even more formidable. Interestingly enough,
Waves MaxxBCL has a new solution to this problem using a very old concept.
Waves MaxxBCL at a Glance:
MaxxBass? Bass Enhancement adds deep bass sound without adding bass frequencies
Waves Renaissance Compressor
L2 Ultramaximizer Peak Limiter
24-bit/96kHz resolution with 48-bit, double-precision processing
Supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced line-level analog signals
MaxxBass Bass Enhancement
Waves' patented algorithm adds stunning bass sound without adding bass frequencies,
delivering a bigger bottom end. MaxxBass? allows your audience to hear bass frequencies
lower than physically present by applying psychoacoustic principles: Even though we
can hear a bass guitar from a small speaker, we don't actually hear the fundamental
frequency because the speaker can't produce a pitch that low. Instead, we hear the
harmonics that the speaker can produce, and this causes the brain to create the "missing
fundamental." MaxxBass? takes this well-known psychoacoustic phenomenon to the
maximum, giving you the ability to extend the perceived frequency response of a system
about two octaves below its physical limitation.
To accomplish this, the signal is split: high frequencies are passed to the output (to be
added back to the bass). The bass signal is analyzed and a specific series of upper
harmonics are created. Because the dynamics of the original bass are duplicated in these
harmonics, the result is the most natural sounding bass enhancement available. The
MaxxBass? harmonics and the original bass can be mixed in any proportion at the output.
To provide more control, a high-pass filter can be switched in, allowing the harmonics only to
be passed along as an "image" of the original bass frequencies, which is useful when working
with a system with known low-frequency limitations to avoid over-excursion of the speaker drivers.
Renaissance Compressor
Designed to provide the classic warm sound of analog compressors, the legendary
Renaissance Compressor controls dynamics with studio-style warmth that flatters
full mixes, vocals, or instruments. Controls include selection of vintage-style Opto
or modern Electro compression, and threshold, ratio, and attack controls. Waves
ARC? (Automatic Release Control) algorithm dynamically optimizes the compressor's
release value for a wide-ranging input. ARC? reacts much the way a human ear
expects, and can produce increased RMS level with greater clarity.
In general, the release is faster for peak transients and slower for the overall RMS
level. The ARC? system varies the release time to fit the ear's expectations while
increasing RMS, and without creating distracting artifacts. In this way, the
Renaissance Compressor can serve as a leveler plus a fast compressor simultaneously.
L2 Ultramaximizer Peak Limiter
Heard on countless hit records and soundtracks, the L2 Ultramaximizer puts sound
up-front with breathtaking transparency. The L2 is capable of a very fast, overshoot-free
response. Once the limiter threshold has been set, you can define the actual peak level
that the processed signal will reach. Once set, limiting and level re-scaling becomes a
one-shot process. The L2 can significantly increase the average signal level without
introducing any audible side effects. Yet there is plenty of range to recreate "vintage"
effects such as level pumping or severely limited dynamic range if you like.
Pristine Sound Quality
The MaxxBCL offers 96kHz, 24-bit resolution with a 48-bit, double precision internal
processing path and a dynamic range of ~125dB. The totally passive analog input
path to the ADC uses Jensen transformers, while the output path also features Jensen
analog output transformers. The unit is galvanic-isolated, which prevents ground loops
and allows it to operate in electrically unstable environments. Waves MaxxBCL combines
the highest quality converters available with unequalled processing algorithms to offer you
astonishing new power in bass enhancement, dynamics processing, and format conversion
(analog-to-digital, digital-to-analog, and digital re-quantization).
Versatile Connections
The MaxxBCL supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced
line-level analog signals. MaxxBCL features a unique set of input and output trim settings:
analog input headroom can be set from 9dB above 0dBu to 24dB in six precise steps
(using the rear-panel selectors). Analog output level is similarly set with a separate
rear-panel trim pot offering you the ability to connect between devices with a wide range
of input and output levels.
Waves MaxxBCL Features:
User Interface:
Clear backlit displays, meters, and buttons are visible in all lighting conditions
THD + Noise: ? 0.0006 % @ 1kHz @ -1dBFS
Precision metering with resettable peak hold options (2 sec, infinite)
Accurate, wide-range metering covering 90dB for input and output
and 12dB for compressor and limiter attenuation
Quick access independent bypass on each processing block
Tactile feedback knobs
Four easy store/recall user presets
Input / Output:
Analog, AES/EBU, S/PDIF
Independent rear-panel input and output headroom calibration in six steps
(+9, 12, 15, 18, 20, and 24 dBu)
Passive analog inputs to the A-to-D converters using Jensen analog input transformers
Jensen analog output transformers
High-performance IDR dithering to 16- or 24-bit output
Precise input level setting using 1% resistor networks
Processing:
48-bit end-to-end internal processing path
Switchable compressor/MaxxBass? order
MaxxBass:
Adjustable processor frequency from 25 to 120Hz
Adjustable harmonic mix percentage from 0 to 100%
High-pass Filter "harmonics only" option
Compressor:
Opto/Electro mode selection
Threshold (0-60dB), Ratio (1:1-12:1), and Attack (0.5, 1, 2, 5, 10, 20, and 50 ms) controls
Proprietary ARC? Automatic Release Control
Automatic gain makeup
Limiter:
Overshoot-free look-ahead processing
Adjustable Threshold (0-18dB)
Adjustable Output Ceiling (0-18dBFS)
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GENERAL DATA
Power Supply 220 / 110V 25 VA
Dimension 434x351x44mm (1 rack unit)
Weight » 4 Kg
MEETING INTERFACE IN / OUT
Connector DB 9 female
Nominal in / out audio level 0 dBm
Input impedance 600 W / 10 KW (selectable)
Output impedance 100 W
* nominal line level: - 6 dBm
** it can vary depending on characteristics of telephone line
All measurements are intended at 1 kHz.
AUDIO PROGRAM IN / OUT
Connectors XLR, electr. balanced
Input impedance 600 W / 10K W (selectable)
Output level * - ¥ ¸ + 16 dBm
Output impedance 100 W
Noise on Receive output
2-Wire separation £ 25 dB**
4-Wire separation £ 70 dB
2-WIRE SECTION
Connector RJ11
Nominal input level - 6 dBm
Nominal output level - 6 dBm
Compensation mode Electronic, transf. decoupled
Impedance 600 W
4-WIRE SECTION
Connector Terminal block
Impedance 600 W
Nominal TX level 0 dBm
Nominal RX level 0 dBm
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In today's recording and sound reinforcement environments, the need for multiple channels of high quality, easy to use compression is growing rapidly. The new dbx® 1046 is designed to provide the audio professional with just that: 4 channels of great sounding dbx compression for a variety of applications. Incorporating the industry standard dbx designs and the latest available manufacturing techniques, the dbx 1046 provides pristine sonic quality with that classic dbx sound.
The 1046 provides 4 channels of smooth classic dbx OverEasy® or Hard Knee compression that are perfectly suited for use on individual tracks of your multitrack recorder, and in virtually all applications the separate channels can be individually interfaced and used for entirely independent purposes. Additionally the newly developed PeakStopPlus™ is ideal for protecting your system from the oppressive peaks that can take out valuable drivers in your sound reinforcement rig or studio monitors.
All four channels have the following controls:
Threshold - allows you to set the level at which the compressor starts affecting the gain
OverEasy - allows you to select between soft compression for overall gain control or Hard Knee compression based on the characteristics of the original dbx 160.
Ratio - allows you to set the slope of gain reduction affecting the signal over the threshold level.
Input/Output Meter - allows you to set the meter to check the input and output levels for maximum signal-to-noise ratio and best level matching.
Output Gain - allows you to add make-up gain or to adjust the output level for that channel to match the next device's input gain.
Bypass - hard wire bypass allows you to hear how the compressor is affecting your signal.
PeakStopPlus - allows you to set the maximum signal level you want to pass through this channel. While it's virtually impossible to eliminate distortion, the PeakStopPlus circuit does it gracefully and effectively with minimal distortion.
Stereo Link - allows you to link channels 1&2 and 3&4 for two channels of true stereo compression.
All four channels feature balanced gold plated XLR and 1/4" inputs and outputs, and switchable +4dBu or -10dBV operating level to interface each individual channel with any other device.
So whether you need to control the level, placement in the mix, or overall characteristics of 4 independent signals or control the gain leveling on a couple of stereo pairs, the dbx 1046 is for you.
- Four independent channels of operation, stereo linkable in two pairs
- PeakStopPlus™ limiting control for setting maximum allowable level regardless of compressor settings
- Independent Threshold and ratio controls
- Switchable OverEasy® or Hard Knee compression
- Classic dbx® compression
- Differentially balanced gold-plated XLR and 1/4" inputs and outputs
- True RMS level detection
- Precision metering of input level, output level, and gain reduction
- Dual True stereo or quad mono operation
- Switchable +4dBu or -10dBV operation per channel
Input Connectors | XLR and 1/4" TRS (Pin 2 and tip hot) | Input Type | Electronically balanced/unbalanced, RF filtered | Input Impedance | Balanced > 40 kOhm, unbalanced >20 kOhm | Max Input | > +22 dBu balanced or unbalanced | CMRR | Typically >50 dB at 1 kHz | Output Connectors | XLR and 1/4" TRS (Pin 2 and tip hot) | Output Type | Servo-balanced/unbalanced, RF filtered | Output Impedance | Balanced 30 Ohm, unbalanced 15 Ohm | Max Output | > +22 dBm balanced, > +20 dBm unbalanced | Bandwidth | 20 Hz to 20 kHz, +0/-0.5 dB | Frequency Response | 0.35 Hz to 90kHz, +0/-3 dB | Noise | | Dynamic Range | > 118 dB, unweighted | THD+Noise | >0.008% typical at +4 dBu (1 kHz unity gain), 0.08% typical at +20 dBu (1 kHz unity gain), | IMD | | Interchannel Crosstalk | | Stereo Coupling | True RMS Power Summing | Compressor Threshold Range | -40 dBu to +20 dBu | Compressor Ratio | 1:1 to infinity:1 | Compressor Threshold Characteristic | Selectable OverEasy® or hard knee | Compressor Attack/Release Characteristic | AutoDynamic™ | Compressor Attack Time | Program-dependent | Compressor Release Time | Program-dependent | Compressor Output Gain | -20 to +20 dB | Limiter Threshold Range | 0 dBu to +24 dBu (off) | Limiter Ratio | infinity:1 | Limiter Type | PeakStopPlus® two-stage limiter | Limiter Stage 1 | PeakStop® brickwall limiter | Limiter Stage 1 Attack Time | Zero | Limiter Stage 1 Release Time | Zero | Limiter Stage 2 | Predictive intelligent program limiter | Limiter Stage 2 Attack Time | Program-dependent | Limiter Stage 2 Release Time | Program-dependent | OverEasy® Switch | Activates the OverEasy compression function | I/O Meter Switch | Switches between monitoring input and output levels on the Input/Output Level meter | Bypass Switch | Activates the direct input-to-output hard-wire bypass | Operating Level Switch | (rear panel) Switches the nominal operating level between -10 dBV and +4 dBu simultaneously for both input and output levels | ST Link Switch | Links channels in stereo pairs. Channels One and Three become the master channels | Gain Reduction Meter | 8 segment LED bar graph at 1, 3, 6, 10, 15, 20, 25, and 30 dB | I/O Meter | 8-segment LED bar graph at -24, -18, -12, -6, 0, +6, +12, and +18 dBu | PeakStop™ Indicator | 1 LED to indicate PeakStop™ limiting | Function Switches | LED indicator for each front-panel switch | Output Transformer Options | Jensen® JT-123-dbx or JT- 11-dbx, BCI RE- 123-dbx or RE-11-dbx | Operating Voltage | Switchable:100-120 VAC 50-60 Hz or 200-240 VAC 50/60 Hz | Power Consumption | 20 Watts | Fuse | 100-120 VAC:250 mA Slow Blow, 200-240 VAC:125 mA Type T | Power Connector | IEC receptacle | Dimensions | 1.75“Hx19“Wx9“D (4.4cmx48.3cmx20.1cm) | Unit Weight | 5.2 lbs. (2.4 kg) | Shipping Weight | 7.6 lbs. (3.5 kg) |
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Features Four Independent Compressor Channels Channels 1 and 2, 3 and 4 are stereo linkable 7 LED Gain Reduction metering Balanced XLR Inputs and Outputs Sidechain Insert points Intuitive, user friendly layout Flawless performance The sheer amount of effects and signal processors necessary for today's standards of audio production puts a great strain on the available space in equipment racks, both in the studio and on the road.So, ARX would like to introduce the ARX Quadcomp. An all-new upgrade of the classic award-winning ARX Quadcomp?, the very first four channel 1 RU compressor.You?ll notice that we haven?t strayed very far from our original design concept. The Quadcomp is still four variable ratio Compressor/ Limiters neatly housed in a compact all steel 1 RU package. However, all new low noise/low distortion circuitry, better metering plus XLR inputs and outputs put the Quadcomp in a class of its own. More Control, Less Space On the front panel, each channel has the 'industry standard' individual controls for Threshold, Compression Ratio and Output, plus a new 7 LED Gain Reduction display for accurate visual indication of the amount of gain reduction being applied to the program, and an IN/OUT hardwire bypass switch.In addition to this, each channel has a blank numbered panel to write on for easy confirmation of compressor assigns. Precision Circuitry Internally, each compressor uses Class A VCAs and true 2 pole averaging RMS/DC converters for low distortion and accuracy, plus program dependent attack and release time, which automatically determines optimum compressor response. Balanced Inputs and Outputs The rear panel has true differential Balanced XLR Inputs and Outputs for each compressor. Each compressor has a TRS jack Sidechain access insert point, for frequency sensitive compression, De-essing, etc. in conjunction with an external equalizer (such as the ARX EQ260). Compressor pairs 1 and 2, 3 and 4 also have rear panel mounted Stereo Link switches for accurate stereo tracking. When switched IN, the first channel becomes the master and controls all the functions of the second except Output gain. Universal AC Power AC power range is switchable 100 to 120V, or 220 to 240V, and is connected to the unit via a standard IEC connector, with built-in fuse and voltage switch. Whatever the application ? Channel insert, Bus insert, Master Outputs, Monitor outputs, Crossover outputs, the Quadcomp II?s unique High Density design, precision Low Noise circuitry and clean uncluttered layout make it a truly useful audio tool for all applications.
T E C H N I C A L S P E C I F I C A T I O N S Input Impedance Balanced 20 Kohms, Unbalanced 10 Kohms Input Headroom + 20dB CMRR >50dB, 20 Hz?20 KHz Output Impedance Balanced 300 ohms Unbalanced 150 ohms Output Level (Max) + 20 dB Frequency Response 20Hz-20KHz ±0.5dB Signal to Noise ratio -93 dB Unweighted, -98 dB ?A? weighted Distortion .015% THD @ 0dB,1KHz Dynamic Range 108 dB Attack and Release Times Program dependent Metering 7 LED display: ?1, 2, 3, 6, 12, 18, ?24 dB Sidechain Insert Impedance 10 Kohm Power Requirements 100/120 V AC or 220/240 V AC Weight 5 lbs/2.2 Kg Dimensions 19“W x 1?“H x 6“D, 482 x 44 x 155mm Input Connector type Balanced XLR Output Connector type Balanced XLR Sidechain Insert Connector TRS Jack
Front Panel Controls Hardwire bypass IN/OUT switch Threshold, Ratio and Output Gain controls 7 LED Gain Reduction display Marker panel for labelling compressor assigns Stereo link status LEDs Rear Panel Connectors Input and Output balanced XLR connectors TipRingSleeve Sidechain insert connectors Channels 1 and 2, 3 and 4 Stereo link switches IEC AC mains connector with inbuilt fuse and voltage change |
Features
Unique High Density design puts SIX gates in
minimum rack space
Intuitive, 'user friendly' layout
Balanced Inputs and Outputs
Fast, low noise opto-coupler circuitry
Sidechain/Key Input points on each channel
Hardwire Bypass switch on each gate
Flawless performance in any audio environment
The sheer amount of effects and signal processors required for today's
standards of audio production puts a great strain on the available space
in equipment racks, both in the studio and on the road.
So, in order to give engineers more control in less space, we've created
the ARX Sixgate; Six full function noise gates neatly housed in a compact,
all steel 1 RU package, without sacrificing features or quality.
Why Six gates?
Research shows that the majority of noise gates get used on drum kits,
which means using typically 5 to 6 gate channels. Compare this to the
two or maybe 4 channels that most other units have, and you'll see the
reason for the Sixgate's success.And, by using a minimalist approach to
drum gates, the ARX Sixgate has enough gates left over for backing vocals
and other instruments.
Industry Standard controls
Front panel controls for each gate consist of the 'Industry Standard' controls
for Release, Depth and Threshold, plus Red and Green LED displays to indicate
Gate Open or Closed status, and an IN/OUT hardwire bypass switch.
In addition to this, each gate has a blank panel to write on for easy confirmation
of gate assigns. No more pieces of masking tape stuck everywhere!
Ultra Low Noise
Internally, each gate has proprietary ARX ultra low noise high-speed opto-coupler
circuitry with program dependent Attack time, which tracks the incoming signal to
automatically determine optimum gate response.
Balanced Inputs and Outputs
The rear panel features true differential Balanced Inputs and Outputs for each gate,
on insulated TRS jack connectors.Each gate channel also has Individual Key Inputs/
Sidechain access insert points which can be used either for gate control by an external
signal, or for frequency sensitive gating (when used in conjuction with an external
equalizer such as the ARX EQ260).
Universal AC Power
AC power range on the Sixgate is a universal 100 to 120V or 220 to 240V, and is
connected to the unit via a standard 3 pin IEC connector, with built-in fuse and voltage
switch.
The Sixgate's unique combination of High Density design, intuitive 'user friendly' controls,
and clean uncluttered layout make it a truly useful audio tool for all applications.
T E C H N I C A L S P E C I F I C A T I O N S
Input Impedance
Balanced 20 Kohms, Unbalanced 10 Kohms
Input Headroom + 22 dB
CMRR >60 dB, 20 Hz-20 KHz
Output Impedance
Balanced 300 ohms, Unbalanced 150 ohms
Output Level (Max) + 20 dB
Frequency Response 20Hz to 20KHz ±0.2dB
Signal to Noise Ratio
Gate Closed: -95 dB Unweighted, -105 dB 'A' weighted
Gate Open, Depth Minimum: -93.5 dB Unweighted -98dB
'A' weighted
Distortion 0 .01% THD @ 0dB,1KHz
Dynamic Range 125 dB
Attack Time Program dependent using high-speed
opto-couplers
Release Time User variable
Input and Output Connector type Balanced Jack
Sidechain Insert Connector TipRingSleeve Jack
Sidechain Insert Impedance 10 Kohm
Power Requirements 100/120 V AC, 220/240 V AC
Weight 5 lbs/2.2 Kg
Dimensions
19“Wide x 13/4“High x 6“Deep; 482 x 44 x 155mm
Front Panel Controls
Individual Release, Depth and Threshold controls
Hardwire bypass IN/OUT switch
Open/Closed gate status LED
Marker panel for labelling gate assigns
Rear Panel
Balanced Inputs and Outputs, on Balanced jack
connectors
Sidechain Insert/Key Input TipRingSleeve jack on
each channel
AC input connector, with voltage switch and fuse
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OPTIMOD-FM 5500 puts coveted five-band and two-band OPTIMOD processing into a single rack unit package and brings it to you at the most affordable price ever.
Quality sound is what 5500 is all about-sound that attracts audiences by providing a polished, professional presentation regardless of format and source material. Exceptional versatility allows you to adjust the processor's audio texture to brand your audio, knowing that the resulting signature sound will remain consistent, cut-to-cut and source-to-source. Branding builds businesses and no other processors have the consistency to brand your sound like OPTIMOD.
With the 5500, your signature sound is just a preset away. An easy, one-knob Less/More adjustment allows you to customize any factory preset, trading cleanliness against processing artifacts according to the requirements of your market and competitive environment. Full Control gives you the versatility to customize your audio further. And, if you're a hard-core processing expert, you can explore Advanced Control to tweak presets at the same level as Orban's factory programmers. This versatility makes the 5500 a superb choice for any format. Its five-band processing is ideal for any pop music format (even the most competitive and aggressive CHR), while phase-linear two-band processing yields ultratransparent sound for classical, classic jazz, and fine arts formats. Regardless of your choice, 5500's optimized technology ensures unusually high average modulation and coverage for a given level of subjective quality. Unlike many lesser processors, the 5500 handles speech particularly well. It's always clean, even when you process for loudness.
If you're concerned about latency because you need to feed live talent headphones off air, be assured that the 5500's ultra-low-latency (5 ms delay) processing will keep the most finicky talent happy. Or use optimum latency (15 ms delay) processing for the most competitive sound with delay that's still low enough to satisfy most any talent.
Versatility doesn't stop with sound.
The 5500 can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains.
When used in this mode, the 5500 must be driven (usually via an STL) by a full-featured FM audio processor (like Orban's 8600) that incorporates pre-emphasis aware HF limiting and peak control. In both modes, the 5500's stereo encoder helps deliver a transmitted signal that's always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.
It is the ideal choice for network broadcasters who process with Orban's flagship OPTIMOD-FM 8600 at the network origination point and who need a processor at every transmitter to eliminate STL overshoots (using the 5500's stand-alone stereo encoder mode) and/or to process local insertions while also eliminating network STL overshoots (using the 5500's audio processor / stereo encoder mode). Moreover, the 5500's two modes make it easy for large government and network broadcasters to manage its inventory of spares because any 5500 can be used as a stereo encoder with or without audio processing.
Available in both modes, the built-in, defeat-able ITU-BS412 multiplex power controller allows the 5500's output to meet even the most stringent European government regulations.
A 10 MHz frequency reference input allows the stereo pilot tone frequency to be locked to GPS or another high-accuracy frequency standard. This improves the performance of single-frequency networks in areas where coverage of the transmitters overlaps.
The 5500's built-in stereo encoder, AES/EBU digital inputs and outputs, and analog I/O permit hasslefree interfacing to any broadcast plant, whether the 5500 is located at the studio or the transmitter. Tight band limiting to 15 kHz means you can use any uncompressed digital STL to pass 5500- processed audio from studio to transmitter without compromising on-air loudness - there's no need to use STL's having 44.1 or 48 kHz sample-rate.
The stereo encoder's stereo sub-channel modulator can operate in normal double sideband mode and in an experimental compatible single sideband mode (SSB/VSB) that is offered to enable users to compare and assess the two modes.
Analog Fallback to Digital control that allows Silence Sense to switch the active input from Analog to Digital if silence is detected in the analog input signal but not on the digital input signal. This function works vice versa as well on both analog and digital AES input. The silence sense parameters apply to both simultaneously and both detectors are available to drive the 5500's tally outputs and sending SNMP Traps/Alerts.
If you want to locate the 5500 away from the studio, you'll be pleased by its three separate remote control ports - GPI contact closures, RS232 serial and built-in Ethernet for TCP/IP networks. The serial and Ethernet ports are supported by the supplied 5500 PC Remote Control application. This Windows® 2000/XP/Vista/7/8 application allows you to do even more with the 5500 than you can do through its front panel, making remote control a pleasure.
5500 PC Remote software allows you to access all 5500 features and allows you to archive and restore presets, automation lists, and system setups (containing I/O levels, digital word lengths, GPI functional assignments, etc.)
Built-in clock-based automation lets you automatically day-part the processing. You can control many other 5500 operating parameters too.
The 5500's feature set fully exploits the processor's DSP and computer-based control architecture. To ensure absolute accuracy, you can automatically synchronize the clock to an Internet timeserver. It has a cool-running, energy-efficient switching power supply and uses the latest dual-core DSP chip technology from Freescale Semiconductor.
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