Weiss

With the DAC50x we are creating a new paradigm for what used to be a black box device. A typical D/A Converter is a "set and forget" device. Not so with the DAC50x. It adds a number of interesting signal processing features and sports a variety of digital inputs. Balanced, unbalanced and headphone outputs are provided.Weiss Engineering has a 30 year history in D/A Converter design. In that time span we have learned a thing or two about converter design. The DAC50x is the essence of our experiences. DAC501 and DAC502 are Roon Ready.The DAC501-4ch and DAC502-4ch versions are now also available. The "4ch" indicates that those units will be 4-channel playback capable besides the standard 2-channel operation. In addition they sport the latest D/A Converter chip for a no-compromise performance. Standard DAC501 and DAC502 units can be upgraded to the DAC501-4ch or DAC502-4ch version respectively.Mechanics
The DAC50x uses a stainless steel chassis with a solid 10 mm aluminium front plate.Power Supply
A powerful non-switching power supply is used. All sensitive voltages have their own regulators which are separated between left and right channels. The result is an analog output free of “digital noise” and channel crosstalk. The power switch activates a semiconductor relay which only switches on or off at zero crossings of the mains voltage. This assures a glitch free power switching. The two mains transformers are toroidal types. Mains voltage selection is done automatically by measuring the mains voltage before power is applied to the rest of the electronics. Synchronization
An internal high precision / low jitter clock generator is responsible for clocking the D/A converter section. The sampling frequency of that generator is fixed at about 195kHz. The input signals are converted to the 195kHz sampling frequency for optimal signal quality. This scheme also helps significantly in reducing any jitter related effects. All standard sampling frequencies up to 384 kHz plus DSD x64 and x128 are supported.
Digital Inputs
There are a total of five inputs:* AES/EBU or S/PDIF via XLR, Toslink and RCA sockets* UPnP / DLNA via Ethernet* USB* Roon ReadyAccepted formats: PCM 44.1kHz up to 384 kHz, DSD 64x / 128x.Future formats can be accommodated for via software updates.Signal Processing
The DAC50x has a digital signal processing chip built in (DSP). The following DSP algorithms are currently implemented:Room Equalizer - to suppress room modes for a decent bass reproduction.Creative Equalizer - a tone control with low boost/cut, high boost/cut and mid boost/cut. Very useful to correct those recordings which do not quite sound right.De-Essing - the automatic removal of overly bright sibilances from human voices. The sibilance effect can be more or less pronounced depending on your speakers or room acoustics.Constant Volume - adjusts the audio volume (loudness) to a constant value across all tracks played. Useful for "party mode" when the volume control should stay untouched.Vinyl Emulation - get that special sonic character of a record player based playback chain. We also employ an emulation of the DMM-CD procedure offered by the Stockfisch label.Crosstalk Cancelling (XTC) - for the playback of dummy head recordings or live recordings via speakers for an incredible live sensation. Dummy head recordings usually are listened to via headphones because they only work properly if the left channel goes to the left ear only and the right channel to the right ear only. With speakers this is difficult to achive as the left channel goes to the left and the right ear. But with some clever signal processing of the speaker channels is is possible to suppress the crosstalk, i.e. the audio going from the left speaker to the right ear and vice versa. If that works properly then the recording sounds as if one would be in the space where the recording has taken place. All the reverberation and 3D representation of the sound sources is there.(For speaker based playback only.)Loudness Control - a listening volume dependent equalization of the audio.Headphone Equalizer - to adapt any headphone to the listener's ears in terms of frequency response.Crossfeed - to emulate a speaker based playback impression on headphones.Converters
Two of the latest 32 bit D/A Converter chips are used. Two D/A conversion channels are used per audio channel, resulting in exceptional performance specifications. Analog OutputsLine out unbalanced on RCA connector.Line out balanced on XLR connector.Headphone out on 1/4" Jack Analogue.For the DAC502: Addtional headphone output on a 4 pin XLR connector.Discrete output stages for both line and headphone outputs are employed.
The output levels can be set in a coarse manner with 4 steps to adapt for the amplifier or headphone connected. The levels can be set independently for line and headphone outputs.
No sound degrading servo mechanisms are used.Front panel controlsA rotary encoder knob for changing parameters and for powering the unit on/off.A touch screen colour LCD display.An 1/4 inch headphone socket.An IR receiver.Back panel elementsAnalog outputs on XLR and RCA connectors.Digital inputs on XLR, RCA, TOSLINK, USB, Ethernet connectors.USB type A connector for various applications.Mains connector with fuses.IR Remote Control
Allows to control several functions of the DAC50x, namely:The input selected for conversion.The output type, level, muting, absolute polarity.Power on/off.DSP presets.Web InterfaceThe DAC50x has a web interface for configuring it via a web browser. The following functions are accessible via the web interface:Volume and Balance controls.Input selection.Output type.DSP algorithms configuration.DSP snapshots configuration.

With the DAC50x we are creating a new paradigm for what used to be a black box device. A typical D/A Converter is a "set and forget" device. Not so with the DAC50x. It adds a number of interesting signal processing features and sports a variety of digital inputs. Balanced, unbalanced and headphone outputs are provided.Weiss Engineering has a 30 year history in D/A Converter design. In that time span we have learned a thing or two about converter design. The DAC50x is the essence of our experiences. DAC501 and DAC502 are Roon Ready.The DAC501-4ch and DAC502-4ch versions are now also available. The "4ch" indicates that those units will be 4-channel playback capable besides the standard 2-channel operation. In addition they sport the latest D/A Converter chip for a no-compromise performance. Standard DAC501 and DAC502 units can be upgraded to the DAC501-4ch or DAC502-4ch version respectively.Mechanics
The DAC50x uses a stainless steel chassis with a solid 10 mm aluminium front plate.Power Supply
A powerful non-switching power supply is used. All sensitive voltages have their own regulators which are separated between left and right channels. The result is an analog output free of “digital noise” and channel crosstalk. The power switch activates a semiconductor relay which only switches on or off at zero crossings of the mains voltage. This assures a glitch free power switching. The two mains transformers are toroidal types. Mains voltage selection is done automatically by measuring the mains voltage before power is applied to the rest of the electronics. Synchronization
An internal high precision / low jitter clock generator is responsible for clocking the D/A converter section. The sampling frequency of that generator is fixed at about 195kHz. The input signals are converted to the 195kHz sampling frequency for optimal signal quality. This scheme also helps significantly in reducing any jitter related effects. All standard sampling frequencies up to 384 kHz plus DSD x64 and x128 are supported.
Digital Inputs
There are a total of five inputs:* AES/EBU or S/PDIF via XLR, Toslink and RCA sockets* UPnP / DLNA via Ethernet* USB* Roon ReadyAccepted formats: PCM 44.1kHz up to 384 kHz, DSD 64x / 128x.Future formats can be accommodated for via software updates.Signal Processing
The DAC50x has a digital signal processing chip built in (DSP). The following DSP algorithms are currently implemented:Room Equalizer - to suppress room modes for a decent bass reproduction.Creative Equalizer - a tone control with low boost/cut, high boost/cut and mid boost/cut. Very useful to correct those recordings which do not quite sound right.De-Essing - the automatic removal of overly bright sibilances from human voices. The sibilance effect can be more or less pronounced depending on your speakers or room acoustics.Constant Volume - adjusts the audio volume (loudness) to a constant value across all tracks played. Useful for "party mode" when the volume control should stay untouched.Vinyl Emulation - get that special sonic character of a record player based playback chain. We also employ an emulation of the DMM-CD procedure offered by the Stockfisch label.Crosstalk Cancelling (XTC) - for the playback of dummy head recordings or live recordings via speakers for an incredible live sensation. Dummy head recordings usually are listened to via headphones because they only work properly if the left channel goes to the left ear only and the right channel to the right ear only. With speakers this is difficult to achive as the left channel goes to the left and the right ear. But with some clever signal processing of the speaker channels is is possible to suppress the crosstalk, i.e. the audio going from the left speaker to the right ear and vice versa. If that works properly then the recording sounds as if one would be in the space where the recording has taken place. All the reverberation and 3D representation of the sound sources is there.(For speaker based playback only.)Loudness Control - a listening volume dependent equalization of the audio.Headphone Equalizer - to adapt any headphone to the listener's ears in terms of frequency response.Crossfeed - to emulate a speaker based playback impression on headphones.Converters
Two of the latest 32 bit D/A Converter chips are used. Two D/A conversion channels are used per audio channel, resulting in exceptional performance specifications. Analog OutputsLine out unbalanced on RCA connector.Line out balanced on XLR connector.Headphone out on 1/4" Jack Analogue.For the DAC502: Addtional headphone output on a 4 pin XLR connector.Discrete output stages for both line and headphone outputs are employed.
The output levels can be set in a coarse manner with 4 steps to adapt for the amplifier or headphone connected. The levels can be set independently for line and headphone outputs.
No sound degrading servo mechanisms are used.Front panel controlsA rotary encoder knob for changing parameters and for powering the unit on/off.A touch screen colour LCD display.An 1/4 inch headphone socket.An IR receiver.Back panel elementsAnalog outputs on XLR and RCA connectors.Digital inputs on XLR, RCA, TOSLINK, USB, Ethernet connectors.USB type A connector for various applications.Mains connector with fuses.IR Remote Control
Allows to control several functions of the DAC50x, namely:The input selected for conversion.The output type, level, muting, absolute polarity.Power on/off.DSP presets.Web InterfaceThe DAC50x has a web interface for configuring it via a web browser. The following functions are accessible via the web interface:Volume and Balance controls.Input selection.Output type.DSP algorithms configuration.DSP snapshots configuration.

The ADC2 is the successor of our renowned two channel ADC1-MK2 A/D Converter. NEW: FIREWIRE option now available! The firewire option is a PCB which plugs into the main board of the ADC2. It supports bidirectional transmission: - analog or digital in from the ADC2 to the computer - from the computer to the ADC2, out on the S/PDIF connector on the back. All sampling rates are supported. There are drivers for Windows and PowerPC based MACs available. The ADC2 uses state of the art A/D chips in our proven ?correlation technique? configuration, which lowers converter imperfections. The analog input stages are kept balanced from the input connectors throughout to the converter chips. A high quality microphone preamplifier is built in as a standard feature. Supported sampling frequencies are 44.1, 48, 88.2, 96, 176.4 and 192 kHz. Output formats are AES/EBU in one or two wire technique, S/PDIF as well as Firewire for a direct connection to computers. Synchronization can be internal or external through AES/EBU or BNC (Wordclock). The built in digital peak limiter allows for setting a generous headroom on the analog inputs and still get a full scale signal at the converterΆs output. A large bar graph shows the level to the A/D input, the output level and the gain reduction in the aforementioned Limiter. The output wordlength can be reduced from 24 to 16 bits with the built in POW-R dithering. It is even possible to have one output running at 24 bits and another one at 16 bits. This feature comes handy when a safety copy to e.g. a DAT has to be made. The analog input sensitivity can be set in 1dB steps via a relais controlled attenuator. An additional gain control is implemented in the DSP chip in the digital domain. Both channel 1 and channel 2 are fully independent, except for the sampling rate and for the dither settings. The AES/EBU sync input can be used as a digital audio input. This allows to limit and / or dither digital audio signals. The peak hold feature can be used to monitor a transfer and check for overloads which may have occured.

The DAC1 is a stereo 24 bit / 96khz D/A converter designed with the aim of keeping an absolutely uncompromised audio signal path. Much detail and thought was spent on the digital input as well as the analogue output stage. Both have in common the purest possible approach in audio design, aspiring for nothing less than excellence. This is coupled with an ergonomic design that gives the user immediate access to all necessary functions, while keeping an uncluttered and thus easy-to-use front panel. This combination makes a truly professional D/A converter catering for the highest expectations. Inputs: There are three digital inputs on XLR connectors, and one on Toslink (optical). The accepted sampling frequencies are 44.1, 48, 88.2 and 96kHz. AES/EBU signals on a single connector are used. Each XLR input is actively routed to a corresponding XLR digital output, allowing monitoring at multiple stages in a digital studio setup.Synchronization:Several signal reclocking schemes are combined for extremely high jitter attenuation, making the DAC1 virtually immune to jitter over a very wide bandwidth.Converters:The correlation technique (using two converters per channel) which was already successfully employed in the ADC1 gives the DAC1 an edge over other D/A converters with equal wordlength and sampling rate specifications, resulting in improved SNR and THD. Outputs: The discrete Class A outputs have a virtually zero Ohm output impedance, but still can drive large loads without stability problems. Output levels can be set between -infinity and +27dBu. The outputs are symmetrical, but do not have any sound degrading servo mechanisms built in. For asymmetrical operation only one leg of the XLR connector (plus ground) is used. Remote: By hooking up an analog potentiometer or fader to the remote connector, the output level can be remote controlled. This level control happens in the digital domain. The input source selection can also be remote controlled.

The Weiss 102 Series split band De-Esser is one of the mastering industy's most highly praised digital products. Now Weiss engineers have taken their De-Esser design and combined it with the best features of the time proven 102 Series Dynamics Processors. The result is the Gambit DS1, a stand alone digital Dynamics Processor with unparalleled performance and sonic integrity.In De-Esser mode the compression band is selectable as a lowpass, bandpass or highpass and extends the functionality beyond de-essing. The crossover filters are linear phase for the highest sonic quality. The full band dynamics processor with soft knee compressor and hard limiter is ideal for program loudness control. Presets keys let you select a bank of factory presets which can't be changed by the user, but which can be copied into normal snapshots and edited from there.Two banks of factory presets are given, de-esser type and compressor/limiter type. Monitor key: Allows to listen to the filtered signal only, i.e. the one being processed by the dynamics section, facilitates the search for offending signals. Bandsplitting: When touching "bandwidth" or "frequency", the soft keys will select which type of band (low, mid, high).The LEDs above the frequency knob show the current status. Frequency knob selects the corner / center frequency. Attack, Release etc.: Attack, release delay and release fast/slow with averaging knobs make up a powerful yet easy to operate timing section for optimum sound treatment. The LEDs between the knobs light depending on the automatically selected release time. Soft knee knob gives a variable soft knee characteristic (from no soft knee to very gentle). The LED above the knob lights if the input level is in the soft knee region. The LED above the threshold knob lights of the input level is above the threshold. Gain makeup allows for zero gain makeup in DeEsser mode and for as much gain makeup as possible in the compressor/limiter mode. Touching the gain makeup knob assigns the softkeys to select the various modes.

The EQ1-DYN-LP contains both the Dynamic and the Linear Phase EQ programs. After power-up of the unit, the user can chose between Dynamic (DYN) and Linear Phase (LP) modes. This allows to use a single unit for a wealth of different equalizing tasks. The workspace storage, the snapshots and the snapshot backups are fully independent between the DYN and LP parts of the unit. This means that all setup data are retained in any case. Users of EQ1-LP and EQ1-DYN units can upload their current snapshot data into the EQ1-DYN-LP via MIDI.        FEATURES Seven identical parametric bands. All seven bands cover the entire audio frequency range. Each band has Boost/Cut, Frequency and Q/Slope knobs. Each band operates in any of the following modes: High shelving, low shelving, peaking, high cut, low cut, bypass. One parameter per knob operation. Seven sets of controls for seven operating bands. Knobs are touch sensitive. LCD display shows detailed parameters of the touched band. Large, backlit Liquid Crystal Display which shows the overall frequency response (calculated in real time) and the detailed parameter values in dB, Hz and Q. A/B compare memory, 128 snapshot bank with two additional banks for back-up. Digital input / output in AES/EBU format on XLR connectors. Dithering to 16, 20 or 24 bits. POW-R dithering in the LP, DYN and DYN-LP models. 128 steps for boost/cut, frequency and Q parameters. Variable slope shelving filters. Very high Q (up to 650) for notching out offending frequencies. M/S mode for independent equalization of M and S channels. M/S encoder / decoder can be configured separately. Also see the article Stereo Shuffling: New Approach - Old Technique by Michael Gerzon Peak meter, over indicators. MIDI control for each parameter. Bypass, overall gain, CH1/2 independent or ganged.