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AD conversion is 28-bit, and DA conversion is 27-bit. This gives a dynamic range of 120dB. The sampling frequency is either 44.1 or 48KHz (selectable by the user).
Up to 32. Each unit can contain up to eight LMY cards. An LMY2-ML card features four (dual A/B switchable) balanced XLR mic/line inputs. An AI8 unit full of LMY2-ML cards will give 16 A and 16 B mic/line inputs. An LMY4-AD card caters for four line inputs, so a full AI8 unit would contain 32 line inputs. These first two types of card can also be mixed together within the same AI8 unit.
An LMY4-DA card houses four line outputs, so an AO8 unit full of these will provide 32 line outputs.
No. Output cards cannot be houses in an AI8 unit, and input cards cannot be slotted into an AO8 unit.
Up to 320 inputs and 192 outputs. That is ten analogue input units and six analogue output units. However, each engine can process up to either 48 or 96 input channels, and 74 output channels. The extra connections can be used for inserts, direct outputs, or alternative inputs to be used with different patches.
One digital I/O unit can interface up to 64 inputs and outputs to the DSP engine. Eight Mini-YGDAI cards (as used in O1V, D24, DME32) can be slotted into each DI/O8 unit, using ADAT, TDIF, AES/EBU or analogue AD/DA formats. These have 24-bit converters.
Yes. Each input unit has a master phantom power on/off switch. Then each individual mic/line input has remote control phantom power via a switch on the console.
Only the LMY2-ML cards (dual A/B switchable mic/line inputs) have adjustable input gain (+10 to -68dB). This is adjusted using rotary encoders on the console. However, all inputs (mic/line, line, and digital) have a digital attenuation control, situated on the 'SELECTED INPUT CHANNEL' strip.
The maximum analogue input level before clipping is 24dBm. There are DIP-switches on the LMY4-DA cards to select the analogue output level at 24, 18, or 15dBm.
With a switch on the console. A/B inputs are individually selected for each pair, not globally selected, so the A input of one pair could be used at the same time as the B input for a different pair.
No. When an LMY card needs to be replaced while the system is running, first un-plug the 68-pin cable connecting the I/O unit to the DSP engine, then switch off the I/O unit and change the offending card.
Yes, in the SYS/W.CLK menu. The currently connected units are shown with information about the type of cards installed, and their signal levels can be monitored.
AD conversion takes no more than 1.5ms, DA conversion takes a maximum of 1.2ms.
Each AI8 and AO8 unit is connected via one 68-pin cable, using D-type (SCSI-style) connectors. Each DIO8 unit is connected with up to four 68-pin cables (two for up to 64 inputs and two for up to 64 outputs).
The pins are straight-connected, but the cable is made up of 34 twisted pairs, with an impedance of 130ohms. Pin 1 is paired with 35, 2 with 36, and so on up to pin 34 with 68.
32 digital audio signals, wordclock, ID signals and control signals are all carried along these cables.
Up to 200 metres with no loss of audio quality.
No. Yamaha does not currently have any MADI interfacing products. However,there are AES/EBU to MADI converters available from other companies, which could be used to interface a MADI equipped multi-track recorder with the DIO8 unit, for example.
With one 68-pin cable, which carries digital audio and word clock information (2-track inputs, talkback, monitor/cue etc.), and two Ethernet cables using BNC connectors carrying control signals. Two of these cables are needed for bi-directional communication because the signal protocol is one way.
Also up to 200 metres away.
Two. They can either be cascaded to provide extra inputs (up to 192 mono and 16 stereo inputs) or used in 'mirror-mode' (where both DSP engines do the same thing) for redundancy. Even with extra inputs, the output capability remains the same.
All mix, stereo, monitor, cue and talkback busses are shared, and are bi-directional. So an output bus can be sent to an output unit connected to either engine, and inputs from both engines can be routed to the same busses. There is no master/slave relationship between the two engines.
There are six 2-track inputs (two switchable between analogue, coaxial or AES/EBU, and the remaining four are AES/EBU only). Two talkback mic inputs (with phantom power), two analogue monitors and a cue output, and two digital stereo outputs are also provided.
This can easily be done by placing a DIO8 unit in the effects rack. All the out-board equipment can be interfaced by the variety of digital and analogue mini-YGDAI cards available for the DIO8 unit. Then two 68-pin cables can be used to connect the DIO8 unit to the DSP engine (not the console), assuming no more than 32 inputs and 32 outputs are needed. Otherwise, four cables should be used. One DIO8 unit will take up just 4u of rack space. If only analogue connections are needed, the DIO8 can cater for up to 16 line inputs and outputs. If that is not enough, then one AI8 unit and one AO8 unit would be needed. This will take up a total of 6u in the rack, and provide 32 line inputs and outputs. Each analogue unit will need one 68-pin cable to connect to the DSP engine.
Using one DIO8 unit can provide up to 64 digital outputs. Two DIO8 units will provide up to 128 digital outputs. Mini-YGDAI cards, which are slotted into the DIO8, can interface with TDIF, ADAT and AES/EBU connections. Mix, matrix and stereo outputs, and direct outputs from any input channel can then be routed to the outputs of the DIO8 units. Output channels on the console can be routed to any number of output connections, so no splitters are needed. The DIO8 units will need to be connected to the DSP engine (not the console) with one 68-pin cable for every 32 channels used.
There are six 2-track inputs on the rear of the console, catering for analogue and digital inputs. The digital inputs have automatic frequency conversion between 44.1 and 48KHz. These inputs could then be routed to any of the stereo input channels on the console. There are also two digital stereo outputs on the rear of the console, which could be sent to a DAT recorder, for example.
Each analogue input unit (AI8) can be connected to as many as three different DSP engines. This means that the Front Of House, monitor and broadcast consoles (for example) could all share the same inputs. It is possible to connect two DSP engines to each analogue output unit (AO8), though only one DSP engine can be routed to the outputs at any one time. A switch on the unit's front panel determines which of the two DSP engines is routed to the outputs.
Two engines can operate in 'Mirror Mode' for redundancy, and be connected to the same input and output units. Then the AO8 units automatically switch engines in the event of a problem.
With the DIO8 units, if only the first four (out of eight) slots are used, then two DSP engines can share the inputs and outputs. If five or more slots are used, then the connections cannot be shared. However, slots 1-4 could be used by a different console than slots 5-8. In fact, there is a switch in the front of each DIO8 unit to select whether port B interfaces slots 1-4 (the same as port A) or slots 5-8.
Only one console has control of the input gain, phantom power, and A/B input. That console is selectable by a switch on the rear of each analogue input unit.
The control surface is powered by an external supply, the PW1D unit. This is 2u in height, and connects via a (KN-27-32S) 27-pin round plug. There is provision for a second power supply, with automatic switching in the event of failure.
The DSP engine has a built in power supply, using a lockable 3-pin (IEC) connector. This power supply is actually under very little strain (only 200W) due to the presence of only digital audio signals, not analogue.
Yes. There is the facility for two sets of connections (68-pin cable and two Ethernet cables) between the control surface and the DSP engine, with automatic switching in the event of a failure.
The Ethernet cables connecting the console to the DSP engine must be connected while the system is switched off. All 68-pin cables can be connected or disconnected while the system is operational. Obviously audio inputs and outputs can also be connected and disconnected while in use.
The DSP1D is ready to work and play audio within 7 seconds of being switched on. The console will take between 20 and 30 seconds to fully boot up.
Yes.
With a basic DSP engine, up to 48 mono inputs, 4 stereo inputs, 48 mix outputs, 24 matrix outputs and 2 stereo outputs can be processed. Each input channel has a direct output available, and each channel (input and output) has an insert available. By adding another input card to the DSP engine, a maximum of 96 mono and 8 stereo input channels become available.
The internal processing is 32-bit minimum (EQ is 44-bit, for example). There are either six or seven different processing cards inside the DSP1D, depending on the number of inputs used. One card manages the DSP engine, one contains all the effects and graphic equaliser processing, one card contains all the console interfacing (talkback, monitor, 2-track input, etc.), one deals with all the patching, one card processes all the outputs, and either one or two cards process all the input channels. (Each input card can process up to 48 mono and 4 stereo inputs).
This will never happen. All of the functions on the console have their own dedicated processors. There is no DSP 'pool' shared by a number of different functions. All facilities are available at all times for all channels.
Diagnostics software will indicate on which card the problem has occurred. It is then a simple operation to remove the front cover of the unit and replace the malfunctioned part. The unit will need to be switched off first.
There is a virtual patch bay in the software, which is accessed by pressing the 'input patch' button. Input channels are listed vertically, and input connections on the input units are listed horizontally. So any physical input can be routed to any input channel on the console. Furthermore, any input connection can be routed to any number of individual input channels. Each channel could then be processed differently if required. Just click with the mouse on the selected square, or press the enter key and use the cursor keys to navigate around the screen, to connect the highlighted input socket to the highlighted input channel.
Again, a virtual patch bay is available, by pressing the 'output patch button'. All the output channels (mix, matrix and stereo) are listed vertically, while the output unit connections are listed horizontally. Any output channel can be routed to any output unit connection. Furthermore, any output channel can be routed to any number of output connections. So an output channel could be routed to a loudspeaker crossover unit and a multi-track recorder (for example) both at the same time.
Every input channel and output channel has an insert point available. The insert point can be moved (pre-eq, pre-comp, pre-fader for example), and its position can be viewed on one of the 'input patch' or 'output patch' screens. There is an 'insert' button on the selected channel strips (on the console), which enables the insert return. (The insert send is always operational). There is an LED status indicator for the insert on each channel, positioned near the input channel faders and the output channel rotary encoders.
All input channels have a direct output available. The position of the direct output in the audio path can be viewed and altered on the same view as the channel insert. The direct output is then patched to an output connection in another view within the 'input patch' menu.
This can be done by first selecting the internal effect unit you wish to use (press the effect menu button), and then selecting the input and output connections required (channel insert, or send from an output mix and return to an input channel for example). Also, this can be achieved in the virtual patch bay described above: the input patch bay includes effect returns; the output patch bay includes effect sends; the insert patch bays contain both effects sends and returns.
In a similar way to the effects. Either use the virtual patch bays, or select the desired GEQ unit in the 'GEQ' menu, and then select the channel into which you wish to insert it.
Yes. All mono input and output channels can be stereo paired. This is done either by holding on a channel's 'select' button while pressing the 'select' button for the adjacent channel, or by 'clicking' on the heart icon in the software for one of the desired channels. When pairing channels, the properties of the odd numbered channel can be copied to the even channel, or the even channel can be copied to the odd channel. Alternatively, channels can be paired in layers (inputs 1-49, 2-50, 3-51 etc.) instead of the usual adjacent pair (1-2, 3-4, 5-6 etc.). When mix or matrix outputs are paired, the left rotary encoder indicates pan and the right rotary encoder indicates signal level.
Fader level, eq, gate, compression are automatically ganged. Pan, delay, and mix-send levels can be ganged if desired. This is done in the software menus for each of these functions.
As a default, all mix sends are switched on, but their levels are set at minus infinity. The 'selected input channel' area of the console displays all the mix send levels for the selected channel. The send levels can be adjusted here, one input channel at a time. Alternatively, the top row of rotary encoders on the input channel strips all show the send levels to the selected mix bus. Different mix busses can be selected to enable send level adjustments from all the inputs, one mix at a time.
Usually, just one mix send is displayed on the rotary encoders right across the console. However, by pressing the 'LOCAL' button in one particular input module (section of twelve mono input channels), that input module is able to display a different mix send to the rest of the console. For example, mix send 1 could be seen on the encoders in the front-left module, while mix send 2 is seen on all the other mix encoders on the console. If 'LOCAL' is selected in each input module, a different mix send could be seen in each of the four input modules.
Yes. This is particularly useful for monitor mixing. When the blue 'mix fader flip' button, which is situated at the front of the 'selected input channel', is pressed (and its red LED lights up), the input fader and mix send rotary encoders swap roles. So then the input faders will adjust the send levels to the selected mix bus. When a different mix is selected, the faders will change accordingly. If the mix is set to 'fixed', then all the faders will be at zero gain, and will not be adjustable.
This must be done in the software, though there is a 'fixed' LED indicator by each 'mix send level' encoder. Pressing the 'Pan/Routing' menu button will display input channels horizontally and mix busses vertically. The mix busses have 'vari/fix' virtual buttons on the left of the display. When a mix bus is in fixed mode, the send level will be unity (or zero gain).
Simply press the pink 'stereo' button on the desired input or mix output channels.
All 48 mix outputs, the two stereo outputs and a stereo sub input.
Firstly, switch on the 'matrix' button for the required output bus. This button is found above each mix bus rotary encoder, or on the 'selected output channel' section of the console. Then, the 'Matrix/St' menu button must be pressed. On the first screen in this menu, the mix busses and the stereo busses are displayed horizontally and the matrix inputs vertically. Each send has its' own send level rotary control. The next page in this menu shows all the send levels to one particular selected matrix. The send levels can be adjusted either with the mouse (tracker pad) or the silver data wheel and the cursor keys.
The DCAs work in the same way as VCAs on an analogue console. DCA stands for digitally controlled amplifier. There are twelve DCAs available. They can all control the fader levels of input channels. Also, faders 9-12 can be used to control mix bus output levels. The assigned channel faders will remain static even though the level is being changed digitally by the DCA faders.
Either press the desired DCA buttons on the selected channel strip for each channel in turn, or press the yellow 'DCA assign' button for an individual DCA fader, and select the channels to be controlled with the yellow 'DCA' buttons above each input channel fader (or rotary encoder in the case of a mix bus output). DCAs can also be assigned in the software. The 'DCA/Mute' menu buttons show DCA assign virtual patch bays for inputs and outputs.
There are twelve mute groups available. Like the DCAs, mute groups 9-12 can be used to control outputs as well as inputs. Groups 1-8 can only control input channels. The mute groups can only be used if the twelve 'direct scene recall' buttons are not needed, as they are duel purpose switches. Channels have to be assigned to the mute groups in the software. The second page in the 'DCA/Mute' menus display virtual patch bays used to assign input and output channels to mute groups. (Incidentally, each DCA has its own mute switch).
Yes they can: by pressing any of the blue buttons to the right of the DCAs (labelled 'fader status'), the DCA faders can then be used to adjust mix output levels; by pressing the white 'in' button in the 'fader status' strip, the DCA faders can then be used to adjust input fader levels. Any bank of twelve inputs can be adjusted, depending on which individual input channel is selected. Lastly, the DCA faders can also be used to control the gain of all the bands in the Graphic Equalisers.
No. This can only be done in the software, in the 'Matrix/St' menu.
All the inputs (stereo and mono) have digital attenuation, phase, pan, delay, compression, noise gate, four-band parametric eq with a high-pass filter, and fader level.
All the outputs (mix, matrix, stereo) have pan, delay, compression, six-band parametric eq, and output level.
Yes. There are separate libraries for input eq, input compression, input gate, input channel (whole selected channel strip), output eq, output compression, and output channel (whole channel strip). Also, eq, gate and compressor settings can be copied from one channel to another by 'drag and drop' in the relevant menu pages showing a global view of eq/comp/gate settings.
Yes. There are four different (globally assigned) key inputs, any one of which can be used by any channel compressor and gate. The key can be any input or output channel, or the compressor can self-key with a side-chain filter.
Yes. All the parameters for input channels and output channels can be copied with ease.
Up to 250msec for input channels and up to 1000msec (1 second) on output channels. Also, a delay of up to 750msec is available on monitor A.
Yes. The channel delays can be set by mili-seconds, metres, feet, number of samples, beats-per-minute, and number of frames (for video sync-ing).
In the PAN/ROUTING software pages, the ratio between left-right and center speaker send levels can be adjusted. The 'STEREO B' bus is used as the centre send control, with a mono output appearing at both its left and right channels.
They are independent stereo-in stereo-out effects, as in the ProR3 effects unit.
No, but every input channel has its own input level meter next to the fader. Also, all the input levels for each input layer can be viewed on the display screen, in the 'meter' menu. Also, there are five selectable metering points in this software page. In this way, one input layer's levels can be viewed on the LCD screen while a different input layer is displayed on the control surface.
The inputs can be metered pre-attenuate, pre-gate, pre-fader, post-fader, and post-on.
The outputs can be metered pre-eq, pre-fader, post-fader, and post-on.
The user can select to monitor (in mono or stereo) any 2-track input, either stereo output (A or B) or any combination of the mix and matrix outputs. Monitor A has a delay (up to 750mS) available, and can be interrupted by cue signals and a communication input (which has an automatic gate with adjustable threshold). The user can also adjust the monitor 'dim' level when using talkback. All these adjustments can be made in the 'Moni/Cue' menu.
No. The two talkback channels have their own routing system. Talkback 1 (socket on console surface) and talkback 2 (mic input on the rear of the console) are added and can be routed to any combination of the mix, matrix, stereo and monitor B outputs. Also, they can be routed to just one (otherwise unused) output socket on any AO8 unit. Both talkbacks have phantom power, dual input level and phase switching. All these adjustments can be made in the 'Moni/Cue' menu, on the 'Talkback' page.
Yes. A sine wave, pink noise, or burst noise can be assigned to any physical output socket and any mix, matrix, or stereo output. The oscillator can be accessed in the 'Moni/Cue' menu.
Yes. This can be viewed in the 'Moni/Cue' menu. There is also a solo button with an integral red LED on the control surface, below the PCMCIA card slots. All input and output channels have a 'solo safe' mode, accessed from the same software page. The cue facility can be set up to cue only one channel at a time, or it can allow the cueing of multiple channels (either inputs or outputs, not both at the same time). Inputs and outputs can be cued pre-fade or post-fade, and DCAs can be cued pre-pan or post-pan.
990 scene memories, all stored internally, in a compact flash memory.
All fader positions, all input and output channel processing, DCA assigning, internal GEQs and effects settings are all stored in the scene memories.
There are 99 memories available for each of the following: channel names; patch settings; pre-amp settings; input eq; output eq; input compression; output compression; input gate; input channel settings; output channel settings; GEQ. Also there are 199 memories for internal effects.
The name, patch and pre-amp libraries can be linked to scene memories. This means the user can define whether these settings should change or not when a certain scene is recalled.
Yes. Cut, copy, paste, insert, clear, undo facilities are all available.
Yes. Every input and output channel, DCA, GEQ, and effect has a 'recall safe' mode. Each channel has an orange LED indicator next to the channel fader (or rotary encoder in the case of mix and matrix outputs). The selected input and output channel strips both have recall safe buttons to activate this facility. Also, the 'Scene' menu has a 'recall safe' software page.
Yes. By default, new scenes will be recalled instantly. However, a fade time of up to 60 seconds can be set to enable a gradual change from one scene to the next.
Panning and fader levels for all the selected input and output channels will cross-fade. Channels can be selected in the 'Fade Time' software page in the SCENE menu. All other parameters and channels will change instantly.
No. All channels selected in the 'Fade Time' software page have the same fade time.
You can select the required scene number using the number keys or the inc/dec keys (situated below the PCMCIA card slots) and then pressing the 'recall' button. Alternatively, a scene can be selected in the software, on the 'memory' page in the 'scene' menu.
Also, there are 12 direct scene recall buttons available. These also double-up as mute groups, so the user should select which one of these functions is needed. Any scene can be assigned to any of the direct recall buttons. This is a one-touch operation.
No there isn't, but one of the eight 'user define' buttons placed above the mouse pad can be used for this purpose. Another one could be used for 'previous scene recall'.
Yes. This is the 'preview' function. When the console is in preview mode (button above the scene number pad on the console), the channel settings for a new scene can be viewed (and changed if necessary) without affecting the audio or the settings for the current scene.
So when previewing a scene, all the channel faders, eq, mix sends etc. will change according to the previewed scene's settings, and when a parameter change is made (moving a fader for example), the audio will not be affected until that scene is actually recalled. Then when the console is taken back out of preview mode, all the settings will return to their previous positions, and control of the audio will be regained.
Yes. Scene memories can be copied to, and from PCMCIA cards. There are two PCMCIA card slots on the console. 32MB is enough memory to store 990 scenes (each scene memory needs 32KB).
No, but scene snap-shot automation with timecode is possible.
At present, only program change facilities are available. Control change commands will be possible in the near future with a software up-grade.
This is not specified at present.
Yes. Timecode can be input at the console or at the DSP engine, or it can be generated internally. Scenes can be recalled and MIDI messages sent at exact times set by the user.
Yes. There are seven different brightness settings available, selected in the 'Utility' menu.
Yes. There are sockets at the front and rear of the console. The keyboard is particularly useful for naming input and output channels.
It is an 800 x 600 pixel SVGA display.
Yes. There is a 15-pin SVGA-compatible socket for this purpose on the rear of the console.
No, this is not possible. However, a computer running the PM1D software can be connected to the console or the DSP engine to display and provide control of functions on different software pages.
Yes. Auto-display facilities are available for eq, delay, gate, compressor, input unit, routing, fader, solo/cue and oscillator. These can be selected in the 'Utility' menu.
These buttons can be used to recall the next or previous scenes, recall a particular effect, select a particular source for the monitor outputs, select a particular software page to view, and so on. Their uses can be defined in the 'utility' menu.
Yes. The software can run with Windows 95 and 98. All parameters can be set and stored on a P.C. It is particularly useful to perform time consuming tasks like channel naming and routing on a P.C. without having to be at the console. Scene and library information can then be copied to the console via a PCMCIA card or via the RS232 port on the rear of the console.
Yes. A computer can be connected to the serial port on the rear of the console or on the rear of the DSP engine to provide remote control either along-side the console or instead of the console in an emergency.
Software can be updated by inserting a PCMCIA card in the slot on the console. It could also be loaded from a computer via the serial (RS232C) port.
