D.A.V. Electronics BG No. 501 The features of this microphone amplifier consist of low noise and low distortion. It offers switched gain controls for high accuracy level setting (4 dB steps), D.I. Input (high impedance), level indication, 26dB pads (switchable), phase reversal and 48v phantom supply (switchable). Frequency response: 10Hz 150KHz ± 0.25dB Mic gain range: 22dB 66dB in 4dB steps. [pad in 4dB 40dB] Distortion: Less than 0.01% Typ, 0.001% 1KHz @ +20dBu o/p Phase reversal: Switchable Noise: 25 microvolts [110dB ref 20 dBu. Bandwidth 10Hz18KHz input loaded 150 ohms, 32dB gain]. Level indication: Red LED +18dB Green LED +8dB Green LED 0dB 48v phantom: Switchable Clipping level: 29 dBu 26dB pads: Switchable Connectors: Gold plated PCB plug Input XLR x 1, Jack x 1 (DI Input) D.I. input: High impedance [1.8 Meg Ohms] Dimensions: A.P.I. 500 series module (Lunchbox) D.I. gain range: 8 > 36dB in 4dB steps Für weitere Informationen, besuchen Sie bitte die Homepage zu diesem Produkt.
The DAV BG 503 is a high-quality three-band EQ for the hugely popular 500-series format.
The DAV BG503 is a single-channel EQ that takes one slot of your 500-series rack. Featuring all stepped frequencies control, the High and Low frequency bands provide +/- 8dB of gain while the Mid frequency band uses a resonant design with up to +/- 12dB and a Q factor of 1.5. Low band frequency steps include 12.5Hz, 25Hz, 50 Hz, 100Hz, 200 Hz, and 400Hz, while the High Frequency band covers 2kHz, 4kHz, 8kHz, 12kHz, 16kHz and 32kHz. The Mid band generously overlaps the LF and HF bands with frequencies including 240Hz, 500Hz, 750Hz, 3kHz, 4kHz and 5kHz.
Following the heritage of Decca Studio and inspired by the DAV BG3, the BG503 boasts electronically balanced input and output and exhibits low noise and low distortion. The result is a great sounding EQ.
DAV Electronics BG No.1This Microphone Amplifier is ideal for location or studio recording. Because of it's compact size it is easily transportable for location recording and is very discrete in the studio or control room.The features of this microphone amplifier consist of low noise and low distortion. It offers switched gain controls for high accuracy level setting, level indication, switchable high pass filters, input pads, split mode channel 1 shares its output with channel 2. Its main qualities are that it is high quality at a low cost.Technical Specifications:Frequency Response:10Hz - 150KHz ± 0.25dBDistortion:Less than 0.01%Typ, 0.001% 1KHz @ +20 dBu o/p Noise:25 microvolts [-110dB ref +20 dBu. Bandwidth 10Hz-18KHz input loaded 150 ohms, 32dB gain].48v Phantom:Switchable26dB Pads:Ch 1 & Ch 2Gain Range:26dB - 59dB in 3dB steps.[pad in 0dB - 33dB]High Pass Filters:22Hz, 33Hz, 68Hz @ 12dB per octave.Phase Reversal:CH 1 onlyClipping Level:+29 dBuDimensions:60 x 115 x 295mm (HxWxD)
1U rackmount high quality 2 channel microphone amplifier with DI Input. Features The features of this microphone amplifier consist of low noise and low distortion. It offers switched gain controls for high accuracy level setting (4 dB steps), D.I. Input (high impedance), level indication, high pass filter (switchable), 26dB pads (switchable), phase reversal and 48v phantom supply (switchable). Specifications Frequency response:10Hz - 150KHz ± 0.25dB Mic gain range:22dB - 66dB in 4dB steps.[pad in -4dB - 40dB]Distortion:Less than 0.01%Typ, 0.001% 1KHz @ +20dBu o/p High pass filter:68Hz @ 12dB per octave (Switchable bothchannels)Noise:25 microvolts [-110dB ref+20 dBu. Bandwidth 10Hz-18KHz input loaded 150 ohms, 32dB gain]. Phase reversal:Both channels (Switchable)48v phantom:Both channels (Switchable) Level indication:Red LED +18dBGreen LED +8dBGreen LED 0dB26dB pads:Both channels (Switchable) Clipping level:+29 dBuD.I. input:Channel 1 High impedance [1.8 Meg Ohms] Connectors:IEC mainsInput - XLR x 2, Jack x 1 (DI Input) Output - XLR x 2 (bal), Jack x 2 (unbal)D.I. gain range:Channel 1 only -8 > 36dB in 4dB steps Dimensions:19" 1U rack mount. Depth = 150mm
More information:Review(SOS) Low noise and distortion [Decca 1977 circuit brought up to date] 5 frequency switched filter allows you to limit above or below set frequency or use in wideband mode. Recovery time switched, 3secs, 1sec, 0.3secs Threshold calibrated in dBu ref. output level [switched 20, 19, 17, 15, 13, 11, 9, 7, 5, 3, 1, 0dBu] LED metering Stereo couple Limiter/compress off switch Post limit gain makeup [0, 1, 2, 3, 4, 6, 8, 10, 12, 14, 16dBu] Overall low noise and distortion. Für weitere Informationen, besuchen Sie bitte die Homepage zu diesem Produkt.
DAV Electronics BG No.6Stereo high quality limiter / compressor.Features: Low noise and distortion [Decca 1977 circuit brought up to date] Recovery time switched, 3secs and 0.3secs Threshold calibrated in dBu ref. output level LED metering Stereo couple Limiter/compress On/off switch Overall low noise and distortion. Input gain control to +20dB.
8-Kanal Mikrophonvorverstärker. Kompakte Bauweise, extrem hochwertige Verarbeitung. Das Schaltungsdesign des BG No.8 basiert auf den legendären DECCA-Vorverstärkern, die heute noch ihresgleichen suchen. Gerasterte Gain-Potis ermöglichen akkurate Aussteuerung, die Level-LEDs lassen sich in der Empfindlichkeit umstellen. Außerdem besitzt der BG No.1 26 dB-Pad-Schalter in beiden Kanälen, 48 V Phantomspeisung, sowie Trittschallfilter. Hergestellt in Handarbeit Frequency Response: 10Hz 150KHz ± 0.25dB Gain Range: 26dB 59dB in 3dB steps. [pad in 0dB 33dB] Distortion: Less than 0.01% Typ, 0.001% 1KHz @ +20 dBu o/p High Pass Filters: 22Hz, 33Hz, 68Hz @ 12dB per octave. Ab: 44 x 485 x 255mm (HxWxD) Für weitere Informationen, besuchen Sie bitte die Homepage zu diesem Produkt.
This Microphone Amplifier is ideal for location or studio recording. Because of it's compact size it is easily transportable for location recording and is very discrete in the studio or control room.Features:The features of this dual channel microphone amplifier consist of low noise and low distortion. It offers switched gain controls for high accuracy level setting (4 dB steps), one mic channel (channel 1), one dedicated D.I instrument input (channel 2) with balanced line level output and buffered output to amp (ideal for mixing live and D.I. guitar for fuller sound), level indication, high pass filter (switchable on channel 1 onyl), 26dB pads (switchable on channel 1 only) and 48v phantom supply (switchable on channel 1 only).Specifications:Frequency Response: 10Hz - 150KHz ± 0.25dBDistortion: Less than 0.01%; Typ, 0.001% 1KHz @ +20 dBu o/p Noise: 25 microvolts [-110dB ref +20 dBu. Bandwidth 10Hz-18KHz input loaded 150 ohms, 32dB gain]. 48v Phantom: Channel 1 only (Switchable)26dB Pads: Channel 1 only (Switchable)D.I. Input: Channel 2 high impedance [1.8 Meg Ohms]D.I. Gain Range: Channel 2 only -4 > 40dB in 4dB steps Gain Range: 22dB - 66dB in 4dB steps. [pad in -4dB - 40dB]High Pass Filters: 22Hz, 33Hz, 68Hz @ 12dB per octave (Channel 1 only)Level Indication: Red LED +18dBu, Green LED +15dBu, Can be pre-set for your operating levelClipping Level: +29 dBuGround Lift: SwitchableConnectors: IEC mains, Input - XLR x 1, Jack x 2 (DI Input), Output - XLR x 2 (bal),Dimensions: 60 x 115 x 295mm (HxWxD)Für weitere Informationen, besuchen Sie bitte die Homepage zu diesem Produkt.
The SIPP allows the engineer to use outboard processors in ‘parallel’ by allowing any desired mix of wet and dry paths using a continuously variable blend control. The source is fed to a single input and then split into two: The dry path represents the original source which can then be blended with the with the ‘wet’ path which consists of the same source routed to a balanced insert point. The user can connect any piece of outboard, or indeed pieces of outboard to the wet path to achieve a variety of effects. The most common application for this technique is in the implementation of ‘parallel compression’. This technique allows low-level parts of a signal to be brought up, while retaining the transients in the source. It works particularly well on drums and vocals as well as across the mix in both mixing and mastering applications. Other common uses include distortion as a parallel effect but in fact any process could prove to be creatively useful. The most essential element of any parallel process is that the phase relationship between the original source and the processed parallel remains intact. Digital Audio Workstations allow users to route signals out via DA and AD converters and mix the returning audio with the original. Many also compensate for the conversion delays induced in doing so. But the process can be subject to what is referred to as ‘inter-sample delays’ whereby the returning audio is subject to delays which are less than the value that can be represented by a single sample. Therefore the returning audio can be say, half a sample out of time, and no amount of compensation or nudging regions in the digital realm can perfectly align the two. Engineers have traditionally avoided this problem by sending a ‘dry’ signal out of a second converter pair and returning it alongside the processed one. The SIPP avoids this problem by doing the splitting and subsequent summing in the analogue domain. It is also a valuable tracking tool, before the signal has ever reached the digital recorder. As the SIPP splits the signal into two, it can also be used as a one to two, or two to four line level splitter. This is achieved by using both the main output and the insert send as outputs, while the parallel process is in bypass. So say for example you were tracking a vocalist using a compressor in the traditional way, but weren’t quite sure that the compression worked everywhere in the song, you could connect the compressor to the insert send, the compressor output to a second AD converter channel and bypass the parallel process. That way you’d be printing both a compressed and uncompressed vocal to the DAW with no need for an external mixing desk. The SIPP’s ability to split a signal can also allow it to be used to feed an equalised signal to a compressors sidechain. The original source is fed to the SIPP’s input. The insert send is connected to an eq, and the return of the eq to a compressor sidechain input. The parallel process is bypassed. The output of the SIPP is the connected to the compressors audio input. Now the sidechain can be eq’d without the need for a separate eq’d source to be sent from the workstation. Equally it can be used as a pair of two to one line mixers by using the standard inputs and insert return points as inputs. The blend controls the adjust the relative balances between the sources. Say for example you wanted to blend snare top and bottom mics together, compress them and record to disk as a single file, you could feed the snare top to the input, the bottom mic to the insert return and blend to taste. The resulting output would feed the compressor. To take this idea a step further, the ouput from channel one could feed the input to channel two, allowing you to parallel compress or distort the blended snare mics. The SIPP can also be used as a high quality stepped attenuator. Say for example you want to run your preamps ‘hot’ for sonic reasons, as many API users like to do, but you don’t want to risk overloading your converters, you can use the SIPP’s stepped gain control to allow you to do this. Just because the parallel path is there, it doesn’t mean you have to use it.Für weitere Informationen, besuchen Sie bitte die Homepage zu diesem Produkt.
769,00 €*
Diese Website verwendet Cookies, um eine bestmögliche Erfahrung bieten zu können. Mehr Informationen ...