Using the 2882 Hardware

2882 Front Panel

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2882 Front Panel

The 2882 front panel provides ten-segment metering for the 8 analog inputs and outputs. The meters are fast PPM peak reading meters with auto-resetting peak holds.

The front panel also provides 2882 system status at a glance:

The Mobile I/O front panel also provides access to the Headphone output and some associated controls. The headphone output jack is a TRS 1/4” jack that provides the Left Channel on the tip, the Right Channel on the ring and the ground return for the two channels on the sleeve. These signals are all ground referred, so they may also be split and fed single-ended (unbalanced) to an external audio device.

The Mute and Dim buttons provide instant access to simple level control for the headphone output. The Mute button provides a quick, tactile “panic switch” which mutes the front panel headphone outputs in case of accidental feedback loops and other audio unpleasantries. The Dim button attenuates the front panel headphone output by 18 dB.

2882 Rear Panel

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2882 Rear Panel
The Mobile I/O rear panel features:

Making connections to the 2882

There are six classes of connections you can make to the 2882 hardware:

  1. Analog Audio
  2. Copper-based Digital Audio
  3. Optical-based Digital Audio
  4. Clock Sync
  5. FireWire
  6. Power

ANALOG AUDIO CONNECTIONS

The analog I/O connections on the Mobile I/O have been engineered for maximum flexibility in that they support both balanced and unbalanced connections with a wide range of input and output levels and a wide range of matching impedances. This means that Mobile I/O handles sources from mic level to line level and from mic impedance to guitar impedance. With that in mind, there are a number of aspects of the design that you should take into account when interfacing with Mobile I/O.

Whenever possible, use balanced interconnects with Mobile I/O. The performance of balanced interconnects is much higher and much more resistant to noise interference and electrical (power) wiring problems. The expense of balanced interconnects is not substantially higher than unbalanced connections, so if the gear that you are interfacing with supports balanced connection — use it. If you cannot utilize balanced interconnects, there are connection schemes that you can use that will maximize performance.

On input, at line level, it is sufficient to simply use standard unbalanced (TS) connections. If you are interfacing with the Mobile I/O XLR inputs, you will need to ensure that pin 3 is grounded in the unbalanced adapter cable. More information about adjusting the input level can be found in the MIO Console software chapter.

The Mobile I/O XLR inputs are all wired pin 2 hot and the 1/4” inputs are wired Tip hot.

TIP: To use the 2882 TRS input with guitar or bass, you can simply use a standard TS guitar cable (patch cord) and it will work fine. However, you can take advantage of the balanced input design of the 2882 to get more noise rejection than you thought possible on a guitar input.

In order to do this, you will need to make a psuedo-balanced telescoping shield guitar cable. This can be constructed with a TRS connector, a TS connector and balanced microphone cable. This cable will treat the guitar as a floating balanced source and provide a telescoping shield from the 2882 ground.

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Telescoping Shield Cable for Instruments

If you want to use the TRS inputs with balanced microphones, you will need an XLR female to 1/4” TRS balanced plug adapter cable. These are available commercially, or you can construct one easily. The connections are Tip to Pin 2, Ring to Pin 3 and Sleeve to Pin 1:

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XLR to Balanced TRS Cable

On output, the situation is a bit more complex. If you are driving an unbalanced load, you will get the best performance by not connecting the ring of the TRS jack to ground. In order to do this, you can simply use a balanced TRS/TRS connector with the unbalanced gear. You can also construct a special cable with a TRS connector and a TS connector. In this cable, you just let the ring of the TRS connector float:

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TRS to TS unbalanced cable

Alternatively, the TS connector can be replaced with an RCA connector for interfacing with gear that has RCA unbalanced interconnects.

MAKING THE 1/4” CONNECTION

When you connect a 1/4” plug to a 2882 jack, insert it straight and firmly, ensuring that the plug is fully inserted into the jack. If the plug is not fully inserted you will get level shifts, phase flips, distortion, or no sound.

To disconnect a 1/4” plug, firmly pull the plug straight out from the connector body. The connectors on 2882 are stiff, so you may have to exert some force to remove the plug.

MAKING THE XLR CONNECTION

When you connect a Male XLR plug to a 2882 jack, ensure that you have aligned the pins with the connector body and insert firmly until the retention tab clicks.

To disconnect the plug, press the metal retention tab flush against the box, and pull the plug from the 2882.

COPPER-BASED DIGITAL AUDIO

2882 supports 2 channels of digital audio over copper-based connections. These connections can be made using either S/PDIF interconnects with the RCA connectors or with AES interconnects using the XLR connectors. Even though only one of the AES or S/PDIF inputs can be active at any given time, you can have different digital sources connected to each of the input connectors at the same time – you use the MIO Console application to select the active input. Audio routed to the digital outputs will be mirrored by both S/PDIF and AES outputs. This allows you to send the same stereo pair to two devices at once.

We recommend that you use the AES interconnect mechanism to establish the digital communication between the 2882 and other digital devices. The jitter and electrical noise tolerance on AES interconnects is substantially better than with S/PDIF interconnects. The AES interconnect standard is equivalent to balanced audio interconnections. If you need to use S/PDIF interconnects, try to use the shortest cables you can and, if possible, use special purpose 75 ohm S/PDIF or video cables.

The RCA connectors used for S/PDIF are friction fit coaxial connectors. When you connect them, ensure that they are fully inserted and tight.

The XLR connectors used for AES are fully locking. When connecting to them, make sure that you align the pins and insert firmly. When you remove the connector, make sure that you release the lock by pressing the lock release button before you pull the connector out of the 2882.

INTEGRATED SRC

Normally, when working with digital audio transport, you must take care to ensure that all devices communicating with one another are synchronized to the same audio clock. While this is still an important consideration with 2882, the hardware provides a special feature to simplify copper-based digital connections to the box. The digital input on 2882 has an optional asynchronous sample rate converter (SRC) that will automatically match the sample rate of the incoming audio to the sample rate of the 2882. This converter is enabled by default and you can disable it in the System section of the MIO Console. If you have synchronized the 2882 to the external source (using any of the extensive synchronization methods provided by 2882), you will generally want to disable the SRC in order to get 24-bit transparent audio transport over the digital input.

OPTICAL-BASED DIGITAL AUDIO

Mobile I/O provides two TOSLink™ connectors on the back panel. One is a transmit connector and the other is a receive connector. These connectors are used with Plastic Optical Fiber (TOSLink) cables to communicate with other devices. The TOSLINK connectors can be used to communicate with either the ADAT® Optical communication protocol or the Optical SPDIF communication protocol. Each port can be indpenedently switched between the two protocols via MIO Console.

The ADAT Optical standard allows a device to transmit 8 channels of 24-bit audio at up to 50kHz along with digital audio clock information.

The Optical SPDIF communication protocol allows a device to transmit 2 channels of 24 bit audio at 96kHz, along with digital audio clock information.

Since Mobile I/O provides direct routing within the box, you can easily configure the unit to work as an ADAT based 8 channel A/D/A. Refer to the chapter on MIO Console for information about configuring the routing.

CLOCK SYNC

Clock sync is a serious consideration in any digital audio system.

If you are recording analog sources with 2882, you can simply use the unit’s high-quality internal clock source to drive the converters. This is the easiest case to deal with.

If you need to interface with other devices digitally or ensure sample accurate sync with video sources, the extensive clock synchronization capabilities of 2882 will prove to be more reliable (and better sounding) than most higher priced alternatives.

There are four different ways to get external clock information into the unit:

  1. Sending a 1x word clock signal into the WC Input BNC.
  2. Sending a 256x word clock signal into the WC Input BNC.
  3. Sending an AES or S/PDIF signal into the Digital input.
  4. Sending an ADAT signal into the Optical Digital input.

The BNC word clock input port is a 75 Ohm terminated coaxial input. It should be driven by a 75 Ohm source driver and interconnected with 75 Ohm coaxial cable. If you do not use proper cabling and source drive, you will introduce reflections on the word clock cable which will propagate jitter into the recovered word clock. This is true whether you use the port as a 1x WC input or a 256x WC input, but becomes more important when the clock signal is 256x.

1x is generally appropriate for use with devices that provide a word clock output. If your device provides a 256x output, you may find that you get better results using that clock signal. The Digidesign® line of Pro Tools® products use 256x as their “ SuperClock™” clocking signal.

The AES recommended procedure for distributing clock is to use an AES clock signal. The AES clock signal is an AES digital audio signal with no audio activity. 2882 only uses the AES preambles for clock recovery, so it is immune to data dependent jitter effects. This means you can reliably use the Digital Input as a clock source with or without audio data.

FIREWIRE

FireWire® is Apple’s registered trademark for the IEEE 1394 High-Speed Serial Bus. FireWire started as an Apple technology to replace a variety of interface ports on the back of the computer. After promulgating a number of closed proprietary technologies in the early days of the Macintosh, Apple determined that open standards were better for the Mac, for the industry, and for Apple itself. On that basis they opened their technology for standardization under the auspices of the Institute of Electrical and Electronics Engineers, Inc. (IEEE), an international organization that promotes standards in the field of electronics. FireWire was standardized as IEEE 1394 and promoted for open licensing in the industry.

The first widespread adoption of the technology was for DV camcorders where space was at a premium and bus powering was not perceived as a real issue since all camcorders have batteries. Sony designed an alternative version of the standard 6-pin FireWire connector that provided 1394-based communication with 4-pins in a much smaller form-factor. This version of the connector sacrificed bus-power support and mechanical stability for reduced space requirements. Sony dubbed this version of IEEE 1394 “i.Link®.” This became the de facto standard in the DV world, and was later added to the IEEE 1394 standard. Both i.Link and FireWire refer to the same underlying standard and are completely interoperable. Obviously, i.Link connectors and FireWire connectors cannot be used together without adapters.

2882 uses the FireWire flavor of the IEEE1394 connector with 6-pins for bus power support. The unit ships with two 6-pin to 6-pin FireWire cables, one that is 0.5 meters long (about 18 inches), and the other 4.5m (about 14.5 feet) long. If you want to use 2882 with a 4-pin FireWire device, you will need to purchase a 6-pin to 4-pin adapter cable. These cables are available from a wide variety of retail sources. If you are using a 4-pin cable to connect any device to the computer with 2882, bus power will not be available.

The 6-pin FireWire connector is polarized by its shape, one end of the connector is pointed. The FireWire ports on 2882 point downwards toward the bottom of the box. It will be very difficult to insert the connector upside down, but it is possible if you force it. If the plug is inserted into the socket upside down, the socket will be destroyed. NEVER FORCE A FIREWIRE CONNECTOR INTO A FIREWIRE SOCKET.

Devices connected to the FireWire bus are autoconfiguring. You do not need to set IDs or DIP switches or in any way configure the devices in order to facilitate communication between devices or to configure the bus.

FireWire devices on the same bus must be connected in a tree structure with no loops. This means that devices can be connected to each other in any order, and any device with multiple ports can act as a chain or a hub for other FireWire devices, but you should never be able to get from one device to another by more than one path. If you construct a loop in the bus, it will not operate properly and you will not be able access some or all of the devices on the bus.

Although you are able to attach devices in any order on the FireWire bus, the order of attachment will have an impact on performance. Most current model FireWire devices support 400 Mbs operation, but many older devices may only support 100 or 200 Mbs operation. These devices act as a bottleneck in the bus and limit the speed of any bus traffic that flows through them. In order to maximize performance, you want to ensure that low speed devices are not used to join high speed devices. In practice this generally means that you should attach your 2882 directly to your computer or through a high speed hub.

To connect 2882 to your computer simply plug a FireWire cable into the 2882 and into the computer. The FireWire bus provides a path for all communication between the computer and 2882 – audio, control and meter data.

2882 audio transport takes advantage of FireWire’s support for isochronous transmission, in which the 2882 can reserve a dedicated amount of bandwidth on the bus for moving audio samples. Since the audio must be transmitted on a regular basis to ensure continuous playback and recording , the isochronous model is perfect.

Control changes and meter data are transmitted using asynchronous transactions on the FireWire bus. This transmission approach makes use of the unreserved bandwidth on the bus and competes with things like FireWire hard disk accesses for time. Under normal circumstances this is completely transparent to the user. If the bus becomes overloaded, you may find that disk accesses and meter updates slow down. If you are experiencing bus overloads, you can always add a second FireWire bus with a third-party FireWire card (PC-Card or PCI card depending on your machine), and offload one or more devices to the second bus.

POWER

One of 2882’s great strengths is the flexibility of its power system. 2882 can be powered from any DC source (including bus power) in the range of 9V to 30V as long as it provides 12 Watts of power. The DC inputs on 2882 are a 2.1mm coaxial power connector, center positive and a 4-pin XLR connector Pin 4 Hot. So if you are powering the unit with a third party power source and it supplies 9V, the power source will have to provide 1.4 amps of current. If you are powering the unit with 12V, the power source will have to provide 1 amp of current, and so on.

The 2882 ships with a world-ready 24 volt, 2 amp power supply. You can plug this supply into any AC power source from 90V to 240V, 50Hz - 60Hz, using an appropriate IEC power cord, and it will supply the proper power to the 2882 on the 2.1mm coaxial power connector. 2882 will automatically supply the extra power to the FireWire bus. This means that the 2882 and its power supply can be used to power other bus-powerable FireWire devices including hard-drives, hubs, and other 2882 units.

Since 2882 is DC powered, you can also power up the 2882 using the FireWire bus or another DC source. The 2882 uses 12 Watts of power, so the device supplying the bus power must be capable of sourcing that much power. Most desktop Macs provide more than enough power for 2882 and one other low power device. Most laptops provide enough power for 2882, but not enough for 2882 and another bus-powered device at the same time. If you are using a Powerbook computer, you should not expect to be able to power both the 2882 and a hard drive from the computer. The power capabilities of individual computers vary, so you will have to test the complete system to determine exactly how much your computer can handle.

If you find that the computer is not capable of powering 2882 or does not provide enough run time, you may want to explore using an external power source with the 2882. Check with Metric Halo for details on different battery power solutions for 2882.

As with all electronic devices, when connecting an external power source to the 2882, you should first connect the power source to 2882 while it is in an unenergized state (e.g. not connected to the mains or switched off). After the connection to 2882 has been made, you should energize the power source.

If you connect an energized power source to the 2882’s 2.1mm power connector you may see a small spark when you make the connection. This is due to surge current and is normal if you connect a power source in this way. While this will not damage the 2882 in any way, to avoid the spark just connect the power connector to 2882 before connecting the power source to the wall.

 2882 Overview

2882 Overview

Thank you for purchasing Mobile I/O™, the ultimate FireWire®–based professional audio interface. Mobile I/O is the world’s first portable modular processing audio interface. Your Mobile I/O provides an array of functions that allow you to record and mix with unprecedented quality – Anywhere, Anytime.

What it is

Mobile I/O is a portable, high–quality, modular FireWire–based multi-format audio converter, interface, and processor for professional audio applications. The 2882 model line is equipped with eight balanced analog inputs (4 on XLR and 4 on 1/4" TRS), two channels of Digital I/O (AES/EBU and S/PDIF), eight channels of ADAT® optical I/O support, and eight balanced analog outputs (1/4" TRS), as well as wordclock in/out and 2 IEEE 1394 FireWire connectors that support 400 Mbs operation. All inputs and outputs are capable of 24-bit/ 96kHz operation.

What it has

What you need to use it

What comes with it

Your Mobile I/O 2882 package contains the following items: If any of these items are missing from your package when you open it, please contact Metric Halo or your dealer immediately for assistance.

 CoreAudio

CoreAudio

About CoreAudio™ Technology

CoreAudio is Apple® Computer’s technology standard for interfacing applications to multichannel audio hardware with professional quality. Apple defined the standard and made it the primary interface for audio in OS X. It provides the mechanism for making high-resolution, multi-channel, low-latency connections between audio hardware and audio applications on Mac OS X.

All Mac OS X computer applications provide support for communicating with audio hardware via CoreAudio. As such, it was the natural standard for Metric Halo to support for interfacing with Mobile I/O. Applications (programs) that communicate with hardware via CoreAudio drivers are called CoreAudio Hosts.

The CoreAudio standard is quite rich and provides a number of places where host applications may or may not properly support the specification. There are some hosts that were implemented early that did not get support for multichannel/multistream devices implemented correctly. Most of these hosts have been fixed or are in the process of being fixed now. If you encounter any problems with specific hosts, please let us know about it – but also please let the host vendor know about it. CoreAudio puts many more requirements on hosts than it does on drivers, so it is very likely that any such problems are in the host.

Metric Halo has done extensive testing with the major CoreAudio hosts, and some of the minor ones, and has worked to ensure maximum compatibility with all hosts. Even if you are using a host that is not specifically discussed here, you are unlikely to encounter problems. If you do, please file a bug report with both Metric Halo and the developer of the CoreAudio host. Please send bug reports to support at mhsecure.com with the subject MIOBUG Report.

CoreAudio Basics

How The CoreAudio Driver Works

The Mobile I/O CoreAudio driver is provided by Mac OS X KEXT. The KEXT is a Mac OS X kernel extension. This extension enhances the Mac OS operating system to provide support for communicating with the Mobile I/O hardware. The Mobile I/O driver is implemented as a KEXT due to the requirements of CoreAudio.

The KEXT is provided in a Mac OS X bundle called “MobileIODriver.kext”. This bundle is installed in the /System/Library/Extensions folder. Since this folder is managed by the system, you will have to have administrator access on the computer to install the driver.

The CoreAudio driver provides the required information for CoreAudio to discover and control Mobile I/O. Once the driver has been installed, CoreAudio will automatically find Mobile I/O units as they are attached to the computer and will publish the availablity of the hardware to all interested CoreAudio hosts.

CoreAudio is inherently a multiclient interface — more than one CoreAudio host can communicate with the hardware at the same time. Mulitple hosts can recieve the audio from a Mobile I/O at the same time, and multiple hosts can send audio to the Mobile I/O at the same time. When multiple hosts send audio to the hardware at the same time, CoreAudio will automatically mix the audio before it is sent to the Mobile I/O.

While this mutlticlient operation is a very cool feature of CoreAudio, and can be very helpful for many operations, you must be careful about unintended interactions. In particular, it is very easy to set up the system such that sounds from programs like email clients and other productivity tools will be mixed into your main audio stream (this happens when you set up the default audio output path so that you can use iTunes with the Mobile I/O). If you are not careful, you can check your mail and have the “Mail recieved” sound printed into the bounce that you are doing in the background.

CoreAudio Transport And Sample Rates

CoreAudio supports a wide variety of audio transport standards. As a practical matter, CoreAudio supports multichannel transport of 24 bit audio via floating point streams at virtually any sample rate. Many hosts only support a small number of sample rates and may not support all of the sample rates that are available with the Mobile I/O hardware.

Channel Names

CoreAudio provides a mechanism for the driver to tell the host the names of the channels. Some hosts do not use this information and “make up” their own names for the channels. This mechanism is not dynamic, so the Mobile I/O driver cannot update the host’s names as you adjust the Output Patchbay router or mixer channel assignments.

In v.5 the driver reports stream names to the CoreAudio host for both input and output. Since all I/O in the Mobile I/O is routed by you, you can refer to the names in your application and in MIO Console — they will match. Inputs are called FW 01, FW 02, , FW 18, and outputs are called DAW 01, DAW 02, , DAW 18.

Channel Enables

CoreAudio supports enabling and disabling audio streams. This is a relatively new feature of CoreAudio, and many hosts do not yet support it. It is not clear exactly how the user would control this functionality at this time. For applications that provide manual control over enabled channels, you will get the best performance if you only enable the channels you need. For applications that automatically maintain the enabled channels, you will get the best performance if you only assign channels to outputs you really want to use, and only record enable channels that you intend to record on.

CoreAudio Buffers

Audio channels are transported individually to the host in buffer sized chunks. The size of the audio buffers has an effect on the CPU load of the audio application, as well as the round-trip latency from input to output when the audio is routed through the host application for monitoring or processing.

Generally, the CPU load increases as the buffer size decreases. On the other hand, the latency decreases as the buffer size decreases.

Since, in general, you want the lowest CPU load and the lowest latency, you will have to make trade-offs.

The mixer engine in Mobile I/O helps substantially with this issue, because for the common critical monitoring configuration (monitoring while tracking external sources), the Mobile I/O mixer removes all of the computer transport latency from the monitor path and allows you to decouple the latency from the buffer size.

In the case that you are trying to perform with a softsynth running in your host, Mobile I/O’s mixer does not help decrease the latency since the signal is being generated on the computer. In this case you’ll want to minimize the output latency by selecting the smallest CoreAudio buffer size possible. This will depend on your computer hardware, the amount of processing you are doing, and the CoreAudio host you use. The MIO CoreAudio driver has been optimized to support extrememly small buffer sizes (down to 32 samples on Intel-based HW) and best-in-industry safety offsets.

Setting The CoreAudio Buffer Size

The methods used to set the CoreAudio buffer size will vary from host to host. Some CoreAudio hosts provide direct controls for adjusting the buffer size and others do not. If your host does not support setting the buffer size directly, you will have to use the host’s default buffer size.

Sample Size

The Mobile I/O CoreAudio driver provides the CoreAudio host with 24-bit samples in 32-bit floating point streams. It is the responsibility of the host to dither the incoming audio to 16-bit samples before recording them, if you record 16-bit. If the host does not dither the samples to 16-bit, they will be truncated by the host when they are recorded. For best recording quality, use your host’s 24-bit recording option.

Clock Sources

The CoreAudio specification provides the capability for hosts to control the hardware clock source. Some hosts provide a user interface to do this, others do not. If the host does provide an interface to do this, you will be able to select one of the Mobile I/O external clock sources directly from the host. If the host does not provide an interface, you will need to use the MIO Console to select the external clock source. If you have selected an external clock source using either the host or the console, you will not be able to control the system sample rate from the computer. MIO Console will automatically reflect the clock source and sample rate set by the host.

 Updating your Firmware

Updating your Firmware


The Mobile I/O is a complex device with a complex DSP–based signal processing and control architecture. One of the major strengths of Mobile I/O’s design is that the operating system of the box can be upgraded at any time by updating the firmware. The firmware provides data to the hardware upon system boot that configures both fundamental aspects of the hardware and the operating system for the box. This data is stored in a memory device on the Mobile I/O motherboard. The data can be updated at any time, but it will be maintained indefinitely, even without any power being applied to the Mobile I/O.

Since the hardware itself can be reconfigured by the firmware, this approach allows Metric Halo to make major enhancements to Mobile I/O without any physical changes to the hardware. In the past we have used software deployed firmware updates to increase the FireWire access speed, provide independent heaphone channels, and improve the converter sound quality over its already exceptional character.

Since the firmware updates exist simply as data, they can be sent to you in a variety of ways, whether via CD, email or download from our website. The MIO Console Application provides a built-in tool to update firmware directly from the console. The following section describes how to use the built-in tool.

You may have had the experience of updating the firmware for your computer in the past. As you may know, this can be a stressful procedure, since there is a moment while the old firmware is being replaced by the new firmware, and if the process is interrupted you may be left with no firmware at all. Metric Halo has addressed this issue with a “safe firmware update” procedure. The Mobile I/O uses a dual-boot procedure. The first boot happens in the first 100ms (about 0.1 seconds) and has been extensively tested. It is smart enough to do two things:

  1. It can boot the secondary boot image
  2. It can update the secondary boot image over the FireWire bus

Actually, the primary boot firmware is much smarter than that. The box is completely functional on the primary boot, but all of the more advanced features of the box are enabled by the second boot. The firmware revision of the primary boot is 1.1.00.

As soon as the primary boot image has booted, it checks the secondary boot image, and if the secondary boot image is installed and not corrupted in any way, the system immediately boots the secondary image. If the secondary image is corrupted or if you have held down the front-panel Mute button during the initial boot process, the Mobile I/O will not boot the secondary image and will stay in “Safety Boot Mode”. This is a mechanism you can use if you install firmware that has problems and you need to back up or install a newer image.

The safe boot mechanism allows you to back out of a firmware update if you find that the new firmware has some problem that was not present in a previous version of the firmware. Metric Halo support may ask you to do this during troubleshooting if you encounter any problems.

In the future, Metric Halo may change the methods or tools used to find, download, and accomplish firmware updates. If the tools change, they will be accompanied by an updated version of this Appendix.

Installing a firmware update

In order to install a firmware update, follow these steps:

  1. If there is an associated driver update, install the new driver:
    • Double click the driver installer package and use the Apple installer to update the MIO Driver package.
  2. Reboot your computer.
  3. Make sure your Mobile I/O is powered up and connected to your computer.
  4. Run the MIO Console.
  5. Be sure that you don’t have any audio apps communicating with the MIO (don’t have any audio apps running during the Firmware update).
  6. Select “Update Firmware…” from the “Utilities” Menu. You will see the following dialog:
    2882.png
    Update Firmware Choose Dialog
  7. Select the new Firmware file to install on your Mobile I/O hardware, and click the Choose button. Firmware files will be supplied by Metric Halo with a name that contains the firmware version number in the following format:

    <firmware_version_number>.miofirmware

    Only valid firmware files will be selectable in the dialog box.

The Console will find your Mobile I/O on the FireWire bus and begin sending commands to it. While the update is taking place, a progress window will displayed on the screen:

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Update Firmware Progress Window
When the firmware update has completed, the progress window will disappear.

After the firmware update has completed, you must cycle power on the MIO Hardware for the new firmware to be used, and you must quit and relaunch MIO Console for the new firmware to be recognized:

  1. Disconnect the updated MIO from Firewire and all power sources for a minimum of 30 seconds.
  2. Quit MIO Console
  3. Optional: Reboot your Mac (you may need to do this if the system does not pass audio properly after the update. This can happen if some application is holding on to a reference to the MIO Driver and the driver is not unloaded by disconnecting the unit from the computer).
  4. Reboot the MIO
  5. Reconnect the MIO to the computer
  6. Relaunch MIO Console
    • If everything has proceded properly, the Mobile I/O will be recognized and the new firmware version will be displayed in the “Box Info” pane in MIO Console.
    • If a problem has occured, the firmware will not be updated and/or the box will have safety booted. In this case, repeat the firmware update procedure again from step 1.

Rolling back your firmware

If you find that you have problems with any given release, you can always go back to a previous release by downloading a package from

http://www.mhlabs.com/

and following the update instructions in that package. Please do not roll back the firmware to an earlier version than was originally supplied with your unit unless instructed to do so by Metric Halo.

MIO Console Key Commands

MIO Console Key Commands

MIO Console supports many key commands that you can use. Often, these key commands are supplied with the intention that you will use them with a third-party HID (Human Interface Device) controller device (for example, a Contour Shuttle Pro), so the key commands may involve many modifier keys. Many these key commands can be changed by you, while some of them are fixed.

To view the editable key commands, you can select the Edit > Edit Key Commands… menu command, or, you can use the ⌘⌥⌃Z (Command + Option + Control + Z) key command.

To edit one of these key commands, simply double click the command in the list and the “Set Key Command” dialog will appear:

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Set Key Command
Type the new key sequence you would like to use and click the OK button to set the new sequence.

The following table lists all of the default key commands that can be edited:

Default MIO Console Key Commands

Command Key Sequence
Close All Documents ⌘⌥W (Command + Option + W)
Close All Floating Windows ⌘⇧⌥W (Command + Shift + Option + W)
Close Front Floating Window ⌘⇧W (Command + Shift + W)
Enable High Power Mode (2882) ⌘⌥⌃P (Command + Option + Control + P)
Hide/Show Command Keys Window ⌘⌥⌃Z (Command + Option + Control + Z)
Reset All Meters ⌘D (Command + D)
Hide/Show Mixer Window ⌘= (Command + =)
Hide/Show Console Window ⌘⌥⌃C (Command + Option + Control + C)
Hide/Show All MIO Console Windows ⌘⌥⌃H (Command + Option + Control + H)
Toggle Graph 'Enable PlugIn Window' ⌘⇧I (Command + Shift + I)
Record Panel: Zoom In Channels ⌘⇡ (Command + ⇡)
Record Panel: Zoom Out Channels ⌘⇣ (Command + ⇣)
Record Panel: Zoom In Timeline ⌘⇠ (Command + ⇠)
Record Panel: Zoom Out Timeline ⌘⇢ (Command + ⇢)
Record Panel: Scroll Channels Up ⇧⇡ (Shift + ⇡)
Record Panel: Scroll Channels Down ⇧⇣ (Shift + ⇣)
Record Panel: Scroll Timeline Right ⇧⇠ (Shift + ⇠)
Record Panel: Scroll Timeline Left ⇧⇢ (Shift + ⇢)
Record Panel: Play ⌘J (Command + J)
Record Panel: Stop ⌘K (Command + K)
Record Panel: Record ⌘L (Command + L)
Switch to/from Mini Controller ⌘⌥⌃F (Command + Option + Control + F)
Volume Up ⌘⌥⌃↑ (Command + Option + Control + up arrow)
Volume Down ⌘⌥⌃↓ (Command + Option + Control + down arrow)
Toggle Dim ⌘⌥⌃D (Command + Option + Control + D)
Toggle Mute ⌘⌥⌃M (Command + Option + Control + M)
Toggle Window Visibility ⌘⌥⌃V (Command + Option + Control + V)
Select Monitor Source 1 ⌘⌥⌃1 (Command + Option + Control + 1)
Select Monitor Source 2 ⌘⌥⌃2 (Command + Option + Control + 2)
Select Monitor Source 3 ⌘⌥⌃3 (Command + Option + Control + 3)
Select Monitor Source 4 ⌘⌥⌃4 (Command + Option + Control + 4)
Select Monitor Source 5 ⌘⌥⌃5 (Command + Option + Control + 5)
Select Monitor Source 6 ⌘⌥⌃6 (Command + Option + Control + 6)
Select Monitor Source 7 ⌘⌥⌃7 (Command + Option + Control + 7)
Select Monitor Source 8 ⌘⌥⌃8 (Command + Option + Control + 8)
Select Monitor Output 1 ⌘⌥1 (Command + Option + 1)
Select Monitor Output 2 ⌘⌥2 (Command + Option + 2)
Select Monitor Output 3 ⌘⌥3 (Command + Option + 3)
Select Monitor Output 4 ⌘⌥4 (Command + Option + 4)
Select Monitor Output 5 ⌘⌥5 (Command + Option + 5)
Select Monitor Output 6 ⌘⌥6 (Command + Option + 6)
Select Monitor Output 7 ⌘⌥7 (Command + Option + 7)
Select Monitor Output 8 ⌘⌥8 (Command + Option + 8)

Some of the key-commands depend upon whether or not you are running MIO Console in legacy mode. In Legacy Mode, there are 5 Panels in the MIO Console window, whereas if you disable legacy mode there are only 5 Panels in the MIO Console window.

Pane Select Key Commands in 2d Expanded Mode

Command Key Sequence
Select Pane: I/O Control ⌘1 (Command + 1)
Select Pane: +DSP ⌘2 (Command + 2)
Select Pane: Recording ⌘3 (Command + 3)

Pane Select Key Commands in Legacy Mode

Command Key Sequence
Select Pane: I/O Control ⌘1 (Command + 1)
Select Pane: Mixer ⌘2 (Command + 2)
Select Pane: Routing ⌘3 (Command + 3)
Select Pane: +DSP ⌘4 (Command + 4)
Select Pane: Recording ⌘5 (Command + 5)

MIO Console also provides many key-commands via the menus; these commands cannot be edited:

Menu Key Commands

Command Key Sequence
MIO Console > Preferences ⌘, (Command + comma)
MIO Console > Hide ⌘H (Command + H)
MIO Console > Hide Others ⌘⇧H (Command + Shift + H)
MIO Console > Quit ⌘Q (Command + Q)
File > Open ⌘O (Command + O)
File > Template ⌘⇧O (Command + Shift + O)
File > Close ⌘W (Command + W)
File > Save ⌘S (Command + S)
File > Save As ⌘⇧S (Command + Shift + S)
Edit > Undo ⌘Z (Command + Z)
Edit > Redo ⌘⇧Z (Command + Shift + Z)
Edit > Cut ⌘X (Command + X)
Edit > Copy ⌘C (Command + C)
Edit > Paste ⌘V (Command + V)
Edit > Select All ⌘A (Command + A)
Recording > Set Record Folder ⌘T (Command + T)
Recording > Set Playback Folder ⌘Y (Command + Y)
Recording > Recording Preferences ⌘R (Command + R)
Mixer > Create New Mono Mixer Strip ⌘⇧N (Command + Shift + N)
Mixer > Create Multiple Mixer Strips ⌘⇧M (Command + Shift + M)
Mixer > Create New Bus ⌘⇧B (Command + Shift + B)
Mixer > Configure Mixer ⌘⇧C (Command + Shift + C)
Mixer > Create New Bus ⌘⇧B (Command + Shift + B)
Mixer > Set Color For Selected Strips ⌘⌥C (Command + Option + C)
Mixer > Delete Selected Strips ⌘⇧D (Command + Shift + D)
Mixer > Channel Strip Meters Post Fader ⌘⇧P (Command + Shift + P)
Window > Show/Hide All Floating Windows ⌘B (Command + B)

MIO Console Help




MIO Console Documentation






 MIOConsole Overview

MIOConsole Overview

MIO Console

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MIO Console

MIO Console is the nerve center of your Mobile I/O. Functioning as a standalone application, MIO Console provides full control of every aspect of Mobile I/O. The console software allows you to rapidly and easily adjust all of the Analog Input and Output channel parameters, system sample rate, Digital I/O source, and system clock source. It also allows you to assign ASIO output channels and hardware input channels to the integrated 80-bit, fully-interpolated, multi-bus, near zero-latency hardware mixer.

On Legacy units (units without a 2d Card Installed), MIO Console’s powerful mixer supports both mono and stereo busses, with solo and mute functions for all input and master channels. The mixer bus outputs are routable to any of the hardware outputs, allowing you to easily create multiple simultaneous mixes for send/return busses and multiple live main and monitor mixes. Applications for these features include: foldback support for multiple performers, separate monitor feeds for studio, tape, and control room, and separate mixes for front of house, archive recording, and monitors for live shows. On 2d Expanded Units, you can take advantage of the new v.5 Mixer which is simultaneously more powerful and easier to use. See the v.5 Mixer Help file for complete details.

MIO Console (in Legacy mode) also contains a patchbay router, which allows you to quickly select the source being fed to any output. The patchbay provides easy configuration of stand-alone operation, mix mults, direct outs and various combinations thereof to suit the needs of the moment.

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MIO Console Output Patchbay

In order to simplify work flow and optimize the extent of system control, MIO Console supports comprehensive preset management on both a global and individual control level.

The preset management popup controls within MIO Console allow you to configure various aspects of Mobile I/O and save that configuration information for later recall. Various applications include storing routing configurations for monitor setups, mixer configurations for stem and scene recall, and storing analog level standards for interfacing with external gear and managing different mastering standards.

Global configuration snapshots allow you to save each and every aspect of Mobile I/O's configuration for later, total instant recall. This is useful for preconfiguring Mobile I/O and bringing back the configuration once at the gig, managing separate location setups, or for saving complex studio routing setups for quick changeover.

MIO Console Overview

The MIO Console application consolidates all of the controls for the Mobile I/O hardware into one easy to use window. The Mobile I/O has an extremely large number of user configurable parameters and it is very important that you have instant access to the ones that you need. The console provides a thinly layered interface to the entire system and keeps you from having to deal with “Window Overload”.

The MIO Console window has a view panel selector bar that runs along the top of the window. This bar indicates which of the console view panels is currently active. You can tell which panel is active because the button in the bar is “pushed in.” To switch to one of the other panels, simply click on the name of the panel you want to use. The view will change instantly to the one that you have selected.

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View Panel Pane Selector Bar

Under the view panel selector bar is the currently selected view panel. You control the various aspects of the box with the controls in each view. MIO Console has five main panels:

  1. I/O Panel
  2. Mixer Panel
  3. Routing Panel
  4. +DSP Panel
  5. Record Panel

In v.5 of MIO Console, 2 of these panels are optional, and are only present if Legacy support is enabled. These panels are the Mixer Panel and Routing Panel, which are only needed for working with boxes that have not been expanded with the 2d Card.

I/O Panel

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I/O Panel

This panel provides full control and metering of all of the analog I/O that the box provides. The top half of the view is dedicated to inputs and the bottom half is dedicated to outputs. You access this panel by clicking on the “I/O Control” button of the view panel selector bar.

Box Tabs

At the top of the panel are Box Tabs — one for each box known to the system. You choose which box you are controlling by clicking on the desired box tab. Boxes can be either Online or Offline. If a box is not connected to the computer but it is known to MIO Console, it will be listed as Offline in MIO Console. You can modify the configuration of an Offline box, and that configuration can be saved, but, of course, the changes will not control the Hardware until it is reconnected to the computer.

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Box tabs

MIO Console maintains information about the state of your system persistently. The boxes that have been attached to the computer will be remembered, and their presence in the system can be maintained indefinitely by either the saved system state (saved in your Preferences to maintain the state of the system between launches of MIO Console) or by you explicitly saving the state of the console into a file for later recall (or into a ConsoleConnect host’s session file). In any case, it is possible to have Offline boxes in your system state that refer to a box that you are no longer using. MIO Console provides commands to remove these boxes from the system state.

Each box tab provides a contextual menu that you can popup to apply commands to the specific box. Right-click or control-click the currently active box tab to reveal the popup menu:

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Box Tab Popup Menu

This menu provides the following commands:

ANALOG INPUT CONTROL

For each analog input channel on the Mobile I/O, you will find a channel strip that contains:

  1. Parameter Popup control menu
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    Parameter Pop-up Menu
    • The Parameter Popup control allows you to save, recall, and manage all of the parameters associated with a channel strip in one place. The control is documented in detail later in this chapter (see: “Parameter Popup Controls” on page 50). All presets are automatically shared between all of the input channels.
  2. Phantom Power enable button
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    Phantom Power Button
    • The Phantom Power enable button allows you to control whether or not +48v phantom power is applied to the input by the Mobile I/O. This button is only enabled if you have selected Mic or Mic/Pad as the input level standard. If you are using one of the other level standards, Phantom Power is automatically disabled. If you have enabled Phantom Power on this channel, the button will be illuminated bright red. In addition, the Mobile I/O front panel will illuminate the “Phantom” indicator if any of the input channels have Phantom Power enabled.
    • Phantom power is appropriate for use with condenser microphones or other devices that can (and must) be powered by the preamp. Mobile I/O limits the amount of phantom power to 10mA per channel, preventing device damage due to shorts. Some (rare) microphones may require more power than is provided by 10mA; you will need an external power supply to power those mics.
  3. Level Standard popup menu
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    Level Standard Pop-up Menu
    • This control allows you to select the input level and impedance characteristics for the input channel. The available choices are:
      • Line +4 — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, and the inline pad is engaged. Phantom power is not available. This format is appropriate for interfacing with professional audio equipment.
      • Line -10 — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, and the inline pad is engaged. Phantom power is not available. This format is appropriate for interfacing with prosumer and consumer audio equipment.
      • Inst (Instrument) — This format supports input levels up to +6 dBu. The input impedance is approximately 200k, and the inline pad is not engaged. Phantom power is not available. This format is appropriate for interfacing with high output impedance sources like guitar pickups.
      • Mic — This format supports input levels up to +6 dBu. The input impedance is approximately 200k (12k with phantom power engaged) and the inline pad is not engaged. Phantom power is available. This format is appropriate for interfacing with dynamic and condenser microphones on low to mid level sources. You may find that for very low level sources, especially with low-sensitivity microphones, that your SNR improves if you take advantage of a high-quality external mic preamp. External preamps are recommended for certain types of recording and microphones, especially classical recording and ribbon mics.
      • Mic/Pad — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, (6k with phantom power engaged) , and the inline pad is not engaged. Phantom power is available. This format is appropriate for phantom powered condenser mics that put out near-line level signals (often the case when you are using condenser microphones on bass drums or other very loud sources).
  4. Gain Trim knob
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    Gain Trim Knob
    The Gain Trim knob allows you to adjust the analog gain of the input stage over a 40dB range determined by the input standard that you have selected. The gain is indicated in dB relative to the nominal level of the input standard you have selected. The gain changes are smoother near the bottom of the scale, with the steps increasing in size as you reach the +40dB gain limit.
  5. Channel Label
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    Channel Label
    This simply labels which channel is associated with the channel strip. Click in the lable to edit the channel name.
  6. Channel Level Meter
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    Channel Level Meter

    This is a peak reading, high-resolution, fast PPM meter. It shows the post converter level of the input signal of the associated channel.. The peak hold bar indicates the highest level seen on the channel since the last reset. You can reset the hold by clicking on the meter. These meters are simply high resolution versions of the meters shown on the front panel of the box – all the meter data is generated by the Mobile I/O hardware.

OPTIMIZING INPUT LEVELS

The Analog to Digital converters (ADC) in most devices function best when the peak level is around -6 dBFS (lowest distortion, best sound). This is true of the ADCs in Mobile I/O. Since you have full level control of the input with the gain trim knob, you will find that you get the best quality recordings if you try to set the nominal peak level of the input at about -6 dBFS. In addition to providing the best recording quality, it has the added benefit that you will be operating with an extra 6 dB of headroom before clipping. There is no drawback to optimizing your levels in this way, and plenty of benefit.

ANALOG INPUT CHANNEL LINK

In addition to the channel specific controls, each channel pair shares a Link button. When the Link button is engaged, changes made to one channel of the pair will automatically be applied to the other channel of the pair. This is very useful if you are miking with a stereo pair and need to maintain level balance between the two preamps -- with the Link button engaged, the balance is automatic and exact. When you enable the Link button, the control values for the odd channel in the pair will be copied to the even channel of the pair.

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Analog Input Channel Link

DIGITAL INPUT METERS

To the right of the Analog Input control section is the Digital Input Meter section. This group of meters provides level metering for all of the digital inputs on the Mobile I/O. These meters have the same response characteristics as the analog input meters and show you the audio activity on ADAT channels 1-8 and digital input channels 1-2, going from left to right.

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Digital Input Meters

Please note that when the box is clocking at 2x rates (88.2-96k), the ADAT input uses pairs of optical channels to transport the audio, and only channels 1-4 will show activity – channels 5-8 will not display any signal.

For 2d Enhanced boxes the label above the bank of 8 channel meters is a popup control that allows you to select between ADAT optical format and TOSLINK Optical SPDIF for the optical input connector. Click on the control to popup a menu to select between the different modes of operation. The mode is independently selectable for both the input and output.

If you have selected TOSLINK for either the input or output section, you will only see meters for channels 1+2 of the optical digital section. These correspond to the levels that are being transmitted or recieved in the optical SPDIF signal.

For legacy boxes (boxes that don't have a 2d Card) the label above the bank of 8 channel meters is a popup control that allows you to select between ADAT inputs and Firewire returns. Click on the control to popup a menu to select between the different modes of operation.

CANS CONTROLS

The next control block to the right is the Cans (Headphones) control section.

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Headphone Controls

The Headphone level control knob provides about 40dB of analog gain control on the headphone output. The Mute and Dim buttons reflect the state of the hardware mute and dim controls for the front panel headphone output. Engaging one of these buttons will also engage the associated control on the hardware.

When the Mute function is engaged, the headphone output will be muted. When the Dim function is engaged, the headphone output will be padded by 18 dB. The meters in this section show the headphone output pre-mute/dim block.

SYSTEM CONTROLS

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System Controls

The System block provides controls that adjust various system level aspects of the Mobile I/O hardware:

  1. The Clock Source popup menu controls the system clock source used by the hardware for digital synchronization and driving the converters:
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    Clock Source Pop-up Menu
    • Internal causes Mobile I/O to use its internal clock. You must select this if you want to set the sample rate from the Mobile I/O. If any other clock source has been selected, the console will not allow you to change the sample rate since the sample rate is determined by the external clock source.
    • WC (44/48) directs Mobile I/O to clock off of an external Word Clock Source at single rate (e.g. fs = 32k-50k)
    • WC (88/96) directs Mobile I/O to clock off of an external Word Clock Source at double rate (e.g. fs = 64k-100k)
    • WCx256 (44/48) directs Mobile I/O to clock off of an external 256fs Clock Source at single rate (e.g. fs = 32k-50k)
    • WCx256 (88/96) directs Mobile I/O to clock off of an external 256fs Clock Source at double rate (e.g. fs = 64k-100k)
    • ADAT (44/48) directs Mobile I/O to clock off of the incoming ADAT stream and run at single rate. All 8 ADAT channels are available. You will generally want to select this source if you intend to use ADAT input.
    • ADAT (88/96) directs Mobile I/O to clock off of the incoming ADAT stream and run at double rate. The bottom 4 ADAT channels are multiplexed over the lightpipe to provide 4 channels of double rate audio compatible with other SMUX and Alesis 96k ADAT devices. If you are using a device that provides 4 channels of 96k audio over ADAT optical, you will want to select this clock source.
    • DigIn (44/48) directs Mobile I/O to clock off of the selected stereo digital input at single rate (e.g. fs = 32k-50k). This allows operation of the digital input without SRC, and from devices that must supply clock.
    • DigIn (88/96) directs Mobile I/O to clock off of the selected stereo digital input at double rate (e.g. fs = 64k-100k). This allows operation of the digital input without SRC, and from devices that must supply clock.
  2. The Sample Rate popup menu allows you to select the sample rate when you are using internal clock. The Mobile I/O must be running on internal clock for the Sample Rate popup menu to have any effect. If the Mobile I/O is running from an external clock source, you cannot select the sample rate since it is determined by the external clock source.
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    Sample Rate Popup Menu
  3. The WC Out popup menu allows you to select the output clock signal the Mobile I/O generates on its WC Out BNC connector. The available choices are 1x and 256x. The 1x signal is appropriate for driving devices that accept a Word Clock signal. The 256x signal is appropriate for driving devices that accept 256x or SuperClock signals. Refer to the documentation for the external device to determine what is the most appropriate clock reference for it.
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    WC Out Popup
  4. The DI Source popup menu allows you to select the active input for the digital input pair. The choices are AES and S/PDIF. This selector physically switches the input to the digital audio receiver between the RCA input and the XLR input.
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    DI Source Popup
  5. The DI SRC button enables and disables the asynchronous sample rate converter (SRC) in the Mobile I/O digital audio receiver. When the SRC is engaged (button illuminated yellow), the digital audio receiver will automatically synchronize the input signal to the Mobile I/O system clock over a wide range of sample rate ratios.
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    DI SRC Buton
    This allows you to, for example, digitally transfer a sample from a CD player into a 96k session without any clocking problems. If you want to make bit-transparent transfers, you will need to disengage the SRC and ensure that the Mobile I/O and the external device are both using the same digital audio clock via one of the Mobile I/O synchronization mechanisms.
  6. The Lock indicators show which elements of the Mobile I/O clocking system are properly locked. The clocking system must be locked for the unit to behave as expected. If the system is not locked, audio will play at the wrong rate and will be distorted or noisy. Under normal circumstances, the system should always be locked, but if you have selected an external clock source and the clock signal is not present, corrupted or out-of-range, the system may unlock. There are indicators for the system and the digital input.
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    Lock Indicators

ANALOG OUTPUT CONTROL

The bottom half of the panel is dedicated to the hardware outputs of the Mobile I/O. Much like the analog inputs, each of the eight analog outputs has a set of controls associated with it. The controls are similar to the analog input controls. Each output channel has:

  1. Parameter Popup control — The Parameter Popup control allows you to save, recall, and manage all of the parameters associated with an analog output in one place. The control is documented in detail later in this chapter. All presets are automatically shared between all of the output channels.

  2. Level Standard popup menu — This control allows you to select the output level for the channel. The available choices are:

    • Line +4 — This format supports output levels up to +26 dBu. The standard setting with the trim knob set to 0dB yields a +24dBu output when the digital signal driving the DAC is 0dBFS. This corresponds to +4 dBu nominal with 20dB of digital headroom. You can use the trim knob to adjust the analog output level to be consistent with the needs of other audio gear. The output impedance is approximately 50 Ohms. This format is appropriate for interfacing with professional audio equipment.
    • Line -10 — This format supports a nominal output level of -10dBV with 20 dB of digital headroom. You can use the trim knob to adjust the analog output level to be consistent with the needs of other audio gear. The output impedance is approximately 50 Ohms. This format is appropriate for interfacing with prosumer and consumer audio equipment.

  3. Gain Trim knob — The Gain Trim knob allows you to adjust the analog gain of the output stage (post DAC) over a 40dB range determined by the output standard that you have selected. The gain is indicated in dB relative to the nominal level of the output standard you have selected. The gain changes are smoother near the bottom of the scale, with the steps increasing in size as you reach the +40dB gain limit. Unless the signal is quite low, you will put the output stage into analog clipping well before you hit the 40dB gain limit.

  4. Channel Label — This simply labels which channel is associated with the output.
  5. Channel Level Meter — This is a peak reading, high-resolution fast PPM meter. It shows the pre converter level of the output signal of the associated channel. The peak hold bar indicates the highest level seen on the channel since the last reset. You can reset the hold by clicking on the meter. These meters are simply high resolution versions of the meters shown on the front panel of the box – all the meter data is generated by the Mobile I/O hardware.

ANALOG OUTPUT CHANNEL LINK

In addition to the channel specific controls, each channel pair shares a Link button. When the Link button is engaged, changes made to one channel of the pair will automatically be applied to the other channel of the pair. This is very useful if you are driving a stereo monitor section or stereo device. By engaging the Link button, you will ensure that both channels have precisely the same amount of analog gain applied post DAC. When you enable the Link button, the control values for the odd channel in the pair will be copied to the even channel of the pair.

DIGITAL OUTPUT METERS

To the right of the Analog Output controls is the Digital Meters section. This group of meters provides level metering for all of the digital outputs on the Mobile I/O. These meters have the same response characteristics as the analog output meters, and show you the audio activity on ADAT output channels 1-8 and digital output channels 1-2, going from left to right. Please note that when the box is clocking at 2x rates (88.2-96k), the ADAT output uses pairs of optical channels to transport the audio and only channels 1-4 will show activity, channels 5-8 will not display signal. Both channels of the stereo digital output remain active at all sample rates.

For 2d Enhanced boxes the label above the bank of 8 channel meters is a popup control that allows you to select between ADAT optical format and TOSLINK Optical SPDIF for the optical output connector. Click on the control to popup a menu to select between the different modes of operation. The mode is independently selectable for both the input and output.

If you have selected TOSLINK for either the input or output section, you will only see meters for channels 1+2 of the optical digital section. These correspond to the levels that are being transmitted or recieved in the optical SPDIF signal.

BOX INFO

The Box Info section of the panel, in the lower right-hand corner of the window, shows you information about the currently connected and selected Mobile I/O unit. This section displays the Serial Number, Model Information and Firmware revision of the connected box. All of this information can be useful in trying to track down any connection problems that may arise.

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Box Info

If there is no information displayed in the Box Info section, the software is not communicating properly with the Mobile I/O hardware, or there is no Mobile I/O present on the FireWire bus.

If the FireWire light on the front panel of the Mobile I/O is illuminated but the box information does not appear in the console window, it is very likely that the software has not been installed properly. If this is the case, please refer to the installation instructions for details on how to properly install the software. Also, pleaseensure that there are no copies of the MobileIO Driver in the same folder as the MIO Console application or any of the ASIO drivers that you will use with an ASIO host.

If, on the other hand, the FireWire light on the front panel of the Mobile I/O is not illuminated, the box is not communicating properly with the computer. Please check the cabling of your Mobile I/O and other devices on the FireWire bus and make sure that everything is connected correctly. If that does not properly establish the connection, try rebooting your computer. As a last resort, try connecting only the Mobile I/O to the computer to ensure that communication can be established.

MIXER PANEL — For Legacy Boxes (without 2d Card)

The Analog I/O panel provides all of the bread-and-butter functionality that you would expect from a flexible audio interface like Mobile I/O - what you need to get the job done quickly and easily. The Mixer panel provides features that you won’t find anywhere else – here is where things start to get really interesting.

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Mixer Panel
The controls in the top half of the Mixer Panel are the same analog input controls described in the section I/O Panel. The bottom half of the mixer panel provides a control surface for the multiple integrated hardware mixers running on Mobile I/O. You access this panel by clicking on the “Mixer” button of the view panel selector bar at the top of the console window.

MIXER CONTROLS OVERVIEW

Each mixer has controls for Level (fader), Solo (button), and Mute (button) for each channel. Each stereo mixer also includes a pan control (knob) for each mono channel. For stereo inputs, such as the defaults of the digital inputs or your DAW output, will not have a pan knob. Instead, the two individual channels that make up the pair will be hard panned, and the fader meters will appear as stero meters.

At the far right of each mixer surface is the Master Fader. This fader will be Mono for Mono Mixers and Stereo for Stereo Mixers. Above the Master Fader is a bus mute button that allows you to quickly mute the entire mixer.

Each mix bus can have up to 36 inputs assigned to it. All of the analog, digital, and ADAT hardware inputs are available in each mixer. You can also assign any (or all) of the ASIO playback channels from your DAW to each mixer. Since there are 18 hardware inputs and 18 FireWire return busses that are routed from the ASIO driver on the host computer, you can mix up to 36 channels on each bus.

The default state of the hardware input channels in the mixers is faders at unity, mutes on, solos off and center panned. The playback channels from your DAW are unmuted in pairs per bus (e.g. DAW 1/2 will be unmuted on Mix 1/2, DAW 3/4 will be unmuted on Mix 3/4, etc.) MIXER PANE TABS

Use the tabs above the pan knobs to select the mixer you want to control. Each tab represents a mixer you have configured in the routing panel. When you click on a tab, the controls will be instantly updated to reflect the state of the associated mixer. In this way you can quickly switch back and forth between multiple independent mixes, each with a full mixing interface.

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Mixer Pane Tabs

Each mixer also has a Parameter Popup control associated with it. Unlike the Input and Output controls, the parameter control popup does not have an explicit user interface element representing it. Instead, the popup is accessed from the mixer’s tab itself. To pop-up the menu, you either click and hold the tab, or <control>-click the tab. The parameter popup menu allows you to maintain a library of standard mix configurations, scenes, and setups. It also provides a very quick method for copying mixes from one bus to another.

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Mixer Parameter Popup

CHANNEL FADERS

The Channel fader controls the relative level of the input channel in the mix. It works just like its non-virtual counterpart. The calibration numbers to the left of the fader knob provide an accurate guide to the amount of gain that will be applied by the fader. The label area above the fader knob lets you know which input channel the fader is controlling. The labels default to the console’s default names for the channels, but you can rename the channels to more meaningful names using the naming controls in the Routing Matrix.

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Channel Fader

The exact amount of gain for the fader is displayed in the small black window above the fader channel label. If you want to set a channel gain precisely, simply click in the small black window and type in the desired gain in dB numerically. Hitting the <return> key or clicking outside the entry box makes the new setting take effect. Hitting the <tab> key will make the new setting take effect and will move to the next numeric entry field in the mixer.

The gain range for each channel is from -∞ (muted) to +10 dB. The resolution of the gain control is extremely fine and gain changes are interpolated in the mixer, so there is no zipper noise when you make continuous (or even discontinuous) gain changes. <command> clicking the fader knob will allow you to make fine adjustments to the fader level by dragging the mouse up and down. <option> clicking the fader knob will reset the fader to unity gain. Clicking the meter associated with the fader will clear the peak holds for that channel.

For mono channels, the current gain is applied to the channel before the pan. For stereo channels, the current gain is applied uniformly to both of the hard-panned input channels.

CHANNEL METERS

Each channel fader has a meter (or a pair of meters for stereo channels) associated with it. The meters are calibrated consistently with the fader calibration. Each meter is a peak reading, fast-PPM meter that is pre-fader. The peak hold bar shows the biggest peak since the last reset. Clicking on a meter will clear the peak holds for that channel.

CHANNEL PANS

Each mono input in a stereo mixer has a pan knob above the fader. The pan knob allows you to control the relative amount of the input channel that is placed into the two busses of the stereo mixer. Panning hard left (L100) means that the channel will appear at full volume in the left (odd) bus of the mixer. Panning hard right (R100) means that the channel will appear at full volume in the right (even) bus of the mixer. When the channel is center panned, the signal appears at a decreased volume (-3dB) in both channels, such that the volume of the total signal in both channels is equivalent to full volume in one channel. As you pan from left to right, the signal is distributed between the two channels so that the total volume remains constant.

As with the channel fader, the black window above the channel pan knob provides a precise readout of the current pan position, and clicking on the black window will allow you to type in an exact pan amount. Negative numbers (-100 to 0) indicate left pans and positive numbers (0 to 100) indicate right pans. 0 is center pan. <option> clicking the pan knob will return it to center pan, and <command> click-dragging the knob will allow you to adjust the pan position in finer increments.

MUTE BUTTONS

Each input has a Mute button (labeled “M”) associated with it. When the mute button is engaged (illuminated white), the channel will be muted in the mixer. The mute is interpolated, so muting a channel will not cause audible clicks. <option>-clicking a mute button will set all of the mutes on the mix bus to the same state as the button you click. This allows you to quickly mute all the channels or quickly unmute all the channels on a bus.

SOLO BUTTONS

Each input has a Solo button (labeled “S”) associated with it. When the solo button is engaged (illuminated red), only channels that have been soloed will be mixed by the mixer. The other channels will effectively be muted. As with the mute button, the gain changes associated with soloing or unsoloing a channel are fully interpolated and will not cause audible clicks. <command>-clicking a solo button will exclusively solo the associated channel. The <command>-click will automatically clear the solo state of all other channels on the mix bus. <option>-clicking a solo button will set all of the solos on the mix bus to the same state as the button you click. This allows you to quickly solo all the channels or quickly un-solo all the channels on a bus.

MIXER MASTER FADER

The Fader that appears on the right side of the mixer pane is the Mix Master Fader. This fader controls the overall bus level of the mix bus. Operationally, it is exactly the same as the channel faders (see: Channel F). Each Mix bus in the system has 24 dB of headroom above full scale. If the summing point of the mix bus is clipping, you can pull the mix out of clipping by dropping the Master fader, as long as the sum point is clipping by less than 24 dB.

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Mixer Master Fader

MIXER MASTER MUTE

Each mix bus has a master mute button above the master fader that allows you to mute the output of that mixer. This mute is interpolated.

WIDE MIXERS

As was described before, the integrated mixers in Mobile I/O are WIDE – that is they allow you to mix every available input channel (both hardware channels and FireWire channels) together. In the widest case, the mixer will have 36 faders to allow you to control the gains for all of the input channels.

The MIO console window is only wide enough to accommodate 18 faders plus the Master fader. If you have enabled more channels than will fit in the window, the MIO console will automatically display a scrollbar at the bottom of the Mixer pane. Use this scrollbar to control which faders are visible at any given time. The width of the scrollbar indicator shows you how many of the enabled channels are visible at any given time. The scrollbar will automatically be hidden if you reduce the number of inputs to the mixer below 19 channels or if you switch to a different mixer that has less than 19 channels enabled.

TIP: Mobile I/O’s support of near-zero latency mixing of every channel opens up a huge variety of applications that cannot be achieved with standard interfaces or, at the very least, require external gear or major work-arounds to accomplish. Some examples are:

  • Stem-based mixing.
  • In this mixing technique, you mix disparate elements of the program to separate sub-mixes called stems. You might mix drums to one stem, instruments to another, and vocals to a third. Then the relative balances of the mix can be addressed later in a macroscopic way (during mastering, for example). This also enables remixing the project easily without having to go back to the multi-track master. Since you will be creating stems on individual busses in the DAW, you need to sum the stems for monitoring. This is easily accomplished with Mobile I/O’s mixer.
  • Monitoring mixed-in effects sends when the external effects are unavailable.
  • This is Mobile I/O, right? You may find that you want to continue editing or mixing while you are away from the studio. You can use the WIDE mixer in Mobile I/O to mix in DAW effects send busses for monitoring without having to reconfigure your DAW session.
  • Multichannel foldback mixes.
  • Ever tried monitoring a full drum kit with a host-based foldback mix, or even a 2 channel cue mix? It can’t be done! At least, it can’t be done well). To monitor multichannel input tracking sessions you need an external mixer – or the built in WIDE-mixer in Mobile I/O.
  • If you take advantage of MIO’s ADAT input with a third party A/D converter (or a second MIO) with ADAT outputs, you can simultaneously track and mix up to 18 input channels, with no latency, and no external mixer!
  • Near-Zero Latency monitoring of external effects.
  • Most singers need some reverb or other effects to get the feel right during their performance. With Mobile I/O and the WIDE mixer, you can split off a send from the performer’s input channel, send it to an external effects unit, and mix the effect return into the performer’s foldback mix -- with virtually no latency.
  • Multiple WIDE mixes
  • Since Mobile I/O supports multiple WIDE mix busses simultaneously, you can form multiple, individual foldback mixes for multiple performers at the same time, and best of all, each mix has its own complete mixer control surface, so you don’t have to mess around with a million unreadable aux send knobs.

ROUTING PANEL — For Legacy Boxes (without 2d Card)

The third panel in the MIO Console window is the Routing panel. This panel is what you use to access the powerful routing features that you won’t find anywhere else. The Routing Matrix portion of the panel lets you dynamically control the configuration of the WIDE mixing engine and also configure the channel names for the hardware. You access this panel by clicking on the “Routing” button of the view panel selector bar at the top of the console window.

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Routing Panel

The Routing Panel has two main components:

  1. The Output Patchbay on the left side of the panel.
  2. The Routing Matrix on the right side of the panel.

THE ROUTING MODEL

The Routing Panel provides a user interface for controlling the underlying routing architecture of the Mobile I/O. The routing architecture provides a powerful routing model to allow you to control the routing of signals between physical & virtual inputs and the hardware mixer & physical outputs.

Conceptually, the architecture is quite simple:

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Block diagram of the Mobile I/O routing architecture

All of the physical inputs (e.g. Analog, ADAT and Digital), all of the channels being transmitted over the FireWire bus from the ASIO Application (e.g. DAW), and all of the outputs from the Mobile I/O WIDE Mixer are available to the Output Patchbay. The Output Patchbay can cross-point assign any of its inputs to any of the physical outputs (e.g. Analog, ADAT and Digital).

All of the physical inputs and all of the channels being transmitted over the FireWire bus from the ASIO Application are also inputs to the Mobile I/O WIDE Mixer. Every bus has each of those inputs available for mixing. The mixer outputs are sent to the Output Patchbay for routing to physical outputs. The number of mix busses varies with the Mobile I/O hardware model and the sample rate. The Mobile I/O 2882 supports 10 mono (5 stereo) mix busses at sample rates up to 50kHz (e.g. 1x rates) and 4 mono (2 stereo) mix busses at sample rates up to 96kHz (e.g. 2x rates).

Every physical input is mult’ed from the router/mixer section and sent directly over the FireWire bus to the ASIO host. Regardless of any mixing, routing, or mult’ing that you configure in the hardware, you can always record all of the inputs with your DAW.

As you can see from this simple, high level view, the Mobile I/O routing architecture supports direct routing of any input to any output and also mixing of any set of inputs to multiple mixers. The outputs of the mixers can be routed to any output or any set of outputs for hardware mults. All hardware inputs are available to the ASIO host.

As it turns out, the Mixer/Router in Mobile I/O is quite extensive and the simple user interface hides a lot of complexity:

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Complete view of the Mobile I/O Matrix Mixer

As you can see from this schematic diagram of the mixer/router, the Mixer in Mobile I/O is a fully cross-pointed matrix. While we are presenting the mixer to you as a set of mono and stereo mixers, the actual underlying structure is a full matrix. This means that the mixer is ready for surround and multi-channel matrix mixing today. The schematic provides an accurate representation that shows the signal flow through the mixer and into the router with all analog and digital gain points represented.

OUTPUT PATCHBAY DETAILS

The Output Patchbay allows you choose the signal sources to send to the hardware outputs. You can feed any output from your choice of a mix bus, a hardware input, or a FireWire input from the computer.

PATCHBAY POPUP CONTROLS

The output patchbay is the set of popup menus on the left side of the panel. Each popup menu corresponds to the hardware output that is labelling it. The channel that you choose from the popup menu will drive the associated hardware output.

Since the output patchbay is fully cross-pointed, you can easily construct direct routes from hardware inputs to hardware outputs for stand-alone converter operation, routes from ASIO (using the DAW channels) to the outputs for direct dubbing and monitoring from the computer, and routes from the mixer to hardware outputs for foldback, monitoring, and effects sends.

The output patchbay even lets you send the same channel to multiple outputs, which makes it really simple to create channel mults. You can use this to:

To choose a source for an output channel, simply go to the popup menu for the output you want to route to, click and select the source send. Each popup window is like the jack of a patchbay, and each name in the popup menu is like a patchbay cable that carries the named signal.

PATCHBAY PARAMETER POPUP CONTROL

Directly above the patchbay popup menus is the “patchbay parameters” parameter popup control. This control allows you to store patchbay configurations for later recall. See the section patchbay popup controls, for more details about the parameter popup menu.

The factory default routing for Mobile I/O configures the box to behave as a direct-routed audio interface – all the inputs go directly to the computer, all the outputs from the computer are directly routed to the hardware outputs, and the Headphone output is fed by the stereo mixer on Mix 1/2. This makes the Mobile I/O act like a basic audio interface.

Other configurations that may be very useful include:

This control allows you to maintain a library of frequently used routings and switch between them at will. Spend a little time to familiarize yourself with it — it is extremely powerful.

ROUTING MATRIX DETAILS

The Routing Matrix (Matrix) is the mixer assign matrix that is on the right side of the pane (see “Routing Panel” ). This control surface allows you to fully configure the structure of the multi-bus WIDE mixer in the Mobile I/O hardware. It also allows you to name all of the hardware and virtual channels that are accessible in the hardware.

You use the Matrix to assign hardware and FireWire inputs (DAW outputs) to mixers in the Mobile I/O. To assign channels to a given mix, you click on a crosspoint in the Matrix, darkening the associated Crosspoint Assign tile. Deselecting a tile removes the associated fader from the mixer and mutes the channel in the hardware.

The WIDE mixer allows you to assign all of the hardware input channels and all of the FireWire channels to each mix bus. By limiting the number of channels assigned to any given bus, you can reduce the complexity of the associated mixer interface.

LOGICAL DESCRIPTION

Each mix bus and each input channel has a path label tile associated with it. These tiles are arrayed along the top and left edges of the Matrix.

The tiles along the top edge of the Matrix are the input path tiles. Each tile is color coded based upon the type of input it is:

Each tile shows the Mobile I/O unit it is associated with, the physical name for the path, and the user defined name for the path.

Above each pair of input tiles is a thin tile that is used to join two input paths into a stereo input channel. In the factory default configuration, the Digital Inputs are joined into a stereo pair and each pair of DAW channels is joined into stereo pairs. If the channels are joined into a stereo pair and the pair is assigned to a stereo mixer, the pair will be represented by a single stereo fader in the mixer interface, and no pan knob will appear.

The white tiles along the left edge of the Matrix are the Mix bus tiles. All of these tiles are white because they all represent mix busses. Each tile shows the Mobile I/O unit it is associated with, the physical name for the mix bus, and the user name for the mix bus.

To the left of each pair of Mix tiles are thin inset tiles that are used to join two mix busses into a stereo bus. If the busses are joined, the associated mixer will have pan knobs for each of the mono hardware channels. If the bus is a mono bus, there are no pan knobs associated with the input channels.

The interior of the Matrix is composed of a large number of square crosspoint assign tiles. The Matrix is too wide to fit completely within the MIO Console window; you may use the scrollbar that appears at the bottom of the Matrix to scroll the remaining DAW channels into view.

Each crosspoint assign tile indicates whether or not the input channel at the top of the column is assigned to the mix bus at the edge of the row. If the tile is filled in, the channel is assigned to the mix, and the controls for that channel will appear in the associated mixer. If the crosspoint tile is not filled in, the channel is not assigned to the mix, and the controls will not appear. The channel will be muted and un-soloed on the associated mix bus.

Stereo pairs are automatically assigned as a group to busses.

There are some keyboard shortcuts that you can use when making Matrix assignments:

CONFIGURING CHANNEL NAMES IN THE MATRIX

MIO Console has fully user configurable channel names. The names that you select for your channels will propagate to all of the other aspects of the MIO Console user interface.

This allows you to name the channels in meaningful ways. The analog inputs can be named to match the sources. The Digital I/O can be named to match the effects device that you have patched. Mixes can be named by the foldback monitor or effects send that they will feed.

To name a channel, click on the input path tiles (for input channels) or the mix bus tile (for Mix busses). The channel configuration window will appear above the MIO console window:

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Channel Configuration Window

All of the channel identification controls will be set for the tile you clicked on. To change the user selectable name of the channel, simply type the new name. The name will be updated in the console when you do one of the following things:

You can dismiss the window without updating the channel name by clicking its close box.

You can also select the stereo linking state from Linking popup menu in the channel configuration window. The state will be updated in the console along with the channel name.

The channel identification controls identify which channel you are adjusting:

MATRIX PARAMETER POPUP

The entire state of the matrix can be saved to and recalled from the console parameter library system. The basic functions of the parameter popup control are documented in the next section. Since you are very like to have a number of tracking and mixing configurations you use over and over again, the Parameter Popup for the Matrix is a real timesaver. This control appears in the top left corner of the pane, above the Parameter Popup control for the Patchbay router. Each time you create a configuration that you are likely to use again, save it in the Parameter Library for instant recall when you need it next.

PARAMETER POPUP CONTROLS

The Parameter Popup control is MIO Console’s unified mechanism for handling presets for the various sections of the Mobile I/O. Each element of the console that supports the Parameter Library mechanism has a parameter popup control associated with it. These elements currently include:

Each instance of the Parameter Popup control provides the same commands and options for every section of the console.

POPUP COMMANDS

The parameter popup provides a hierarchical, categorized library of configuration presets for the associated section of the console. The menu is divided into three portions. The first portion consists of all of the items above the “Factory Default” item. The second portion is the “Factory Default” item and the third portion is the hierarchical items below the “Factory Default” item (see Parameter Popup Menu ).

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Parameter Popup Menu

The commands in the first portion of the menu allow you to save and manage the presets in the library. All of the presets are shared between like elements in the console. The preset commands are:

The “Factory Default” command will set the current state of the associated console settings to the default settings.

POPUP PRESETS

In the third part of the menu, each of the categories will be listed as a hierarchical menu title. Each of the presets for each category will be listed in the submenu under the category menu. The currently selected category and preset are drawn in bold, so you will know what is currently active.

Selecting a preset from the menu will make that preset active and will set the current state of the associated console settings to the values contained within the preset. The name of the currently selected preset will be drawn in the popup area in the console window to indicate which preset is active.

If you change the settings in the console, the name of the preset will be drawn in italics indicating that the current settings differ from the selected preset.

For the Input and Output channels, you can hold down the <option> key while selecting a preset to automatically apply the preset to all of the other input or output channels.

To access the parameter popup for the mixers, either click and hold the associated mixer tab or <control> click the associated mixer tab.

We have provided an initial set of presets for the various parameter libraries. The presets for the output channels are relatively complete and give you an idea of the power and flexibility of this approach to parameter management. We will be adding presets on a regular basis – check the website for new presets.

PERSISTENT STATE MANAGEMENT

All Mobile I/O hardware has support for setting a Boot State — the configuration the hardware will use when the unit boots up. As of v.5 of the Mobile I/O software, this boot state includes the entire state of the unit including the configuration of the mixer, the router, sample rate, clocking analog I/O levels (for HW that has digital control), and +DSP configuration.

This functionality allows you to fully configure your hardware and “pour” a complete digital signal processing engine into the HW for instant-on processing.

To configure the Boot State for your Mobile I/O:

  1. First, attach the Mobile I/O to the computer and start up MIO Console.
  2. Use MIO Console to configure the box. Set up all aspects that you care about. Once you have the configuration as you like it, you are ready to save the snapshot.
  3. Choose the “Save Boot State” command from the “Utilities” Menu

The ULN-2 hardware extends the Boot State and adds support for Persistent State Snapshots. There are 10 snapshot slots in the ULN-2 that are recallable from the controls on the ULN-2 front panel. Each Persistent State Snapshot contains a complete description of the state of the box, including Sample Rate, Clock Source, Digital input source, Sample Rate Converter Enable, Patchbay routing, Mixer Configuration, Levels and +DSP configuration and routing. In other words, a snapshot saves every aspect of the configuration of the ULN-2.

The first snapshot slot is special as it is used by the unit to configure the hardware and the routing when the ULN-2 starts up. The other 9 slots are available for storing alternate configurations that can be selected “on the fly” after the ULN-2 is up and running.

When a computer is attached to the ULN-2, the front-panel controls to select snapshots are locked-out since the computer is actively controlling the configuration of the box.

If the computer is not attached, the two tact-switches on the left-side of the front-panel (between the status indicators and the meters) may be used to select the snapshot that you want to use to configure the ULN-2. These buttons are labled with up and down arrows. The currently selected snapshot is indicated by the column of LED’s labled C, 1, 2, 3, 4, 5, 6, 7, 8, 9. When the ULN-2 turns on, the “C” indicator will be illuminated, indicating that the unit has booted up with the state that was stored in the “Boot Snapshot”.

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ULN-2 Front Panel Snapshot Controls

Pressing the up arrow will move to the next higher snapshot in the list (e.g. if you are currently on snapshot 3, you will move to snapshot 2). Conversely, pressing the down arrow will move to the next lower snapshot in the list (e.g. if you are currently on snapshot 3, you will move to snapshot 4). If you are at either beginning of the list and you press the up arrow, you will wrap around to the last item in the list. When you select a new snapshot, the new snapshot is applied to the box immediately.

In order to configure the boot state and snapshots for your ULN-2, you will need to utilize the MIO Console application. Configuring and storing snapshots in the box is very simple:

  1. First, attach the ULN-2 to the computer and start up MIO Console.
  2. Use MIO Console to configure the box. Set up all aspects that you care about. Once you have the configuration as you like it, you are ready to save the snapshot.
  3. Choose the appropriate save command from the Utilities Menu
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    Utilities Menu
    • To save the snapshot to the “Boot State” slot, choose the “Save Boot State…” item.
    • To save the snapshot to one of the other snapshot slots, choose the appropriate “Save Snapshot x State…” item (where x is the appropriate number).
  4. Save a copy of the current Console state to a file on your hard disk with an appropriate name (like “ULN-2 Snapshot 1” for the 1st snapshot) so that you have a copy of the state on the computer if you want to modify it in the future.

With v.5, the 2882 supports an alternate boot state as well. This is the state that is saved in the “Snapshot 1” state slot. If there in nothing saved in this slot, the alternate boot state will be the factory default boot state.

To access the alternate boot-state on the 2882, simply hold the front-panel DIM button while powering the unit. This will select the alternate boot state.

 MIOConsole Overview

MIOConsole Overview

MIO Console

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MIO Console

MIO Console is the nerve center of your Mobile I/O. Functioning as a standalone application, MIO Console provides full control of every aspect of Mobile I/O. The console software allows you to rapidly and easily adjust all of the Analog Input and Output channel parameters, system sample rate, Digital I/O source, and system clock source.

In order to simplify work flow and optimize the extent of system control, MIO Console supports comprehensive preset management on both a global and individual control level.

The preset management popup controls within MIO Console allow you to configure various aspects of Mobile I/O and save that configuration information for later recall. Various applications include storing routing configurations for monitor setups, mixer configurations for stem and scene recall, and storing analog level standards for interfacing with external gear and managing different mastering standards.

Global configuration snapshots allow you to save each and every aspect of Mobile I/O's configuration for later, total instant recall. This is useful for preconfiguring Mobile I/O and bringing back the configuration once at the gig, managing separate location setups, or for saving complex studio routing setups for quick changeover.

MIO Console Overview

The MIO Console application consolidates all of the controls for the Mobile I/O hardware into one easy to use window. The Mobile I/O has an extremely large number of user configurable parameters and it is very important that you have instant access to the ones that you need. The console provides a thinly layered interface to the entire system and keeps you from having to deal with “Window Overload”.

The MIO Console window has a view panel selector bar that runs along the top of the window. This bar indicates which of the console view panels is currently active. You can tell which panel is active because the button in the bar is “pushed in.” To switch to one of the other panels, simply click on the name of the panel you want to use. The view will change instantly to the one that you have selected.

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View Panel Pane Selector Bar

Under the view panel selector bar is the currently selected view panel. You control the various aspects of the box with the controls in each view. MIO Console has five main panels:

  1. I/O Panel
  2. +DSP Panel
  3. Record Panel

In v.5 of MIO Console, 2 of these panels are optional, and are only present if Legacy support is enabled. These panels are the Mixer Panel and Routing Panel, which are only needed for working with boxes that have not been expanded with the 2d Card.

I/O Panel

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I/O Panel

This panel provides full control and metering of all of the analog I/O that the box provides. The top half of the view is dedicated to inputs and the bottom half is dedicated to outputs. You access this panel by clicking on the “ Analog I/O Control” button of the view panel selector bar.

ANALOG INPUT CONTROL

For each analog input channel on the Mobile I/O, you will find a channel strip that contains:

  1. Parameter Popup control menu
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    Parameter Pop-up Menu
    • The Parameter Popup control allows you to save, recall, and manage all of the parameters associated with a channel strip in one place. The control is documented in detail later in this chapter (see: “Parameter Popup Controls” on page 50). All presets are automatically shared between all of the input channels.
  2. Phantom Power enable button
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    Phantom Power Button
    • The Phantom Power enable button allows you to control whether or not +48v phantom power is applied to the input by the Mobile I/O. This button is only enabled if you have selected Mic or Mic/Pad as the input level standard. If you are using one of the other level standards, Phantom Power is automatically disabled. If you have enabled Phantom Power on this channel, the button will be illuminated bright red. In addition, the Mobile I/O front panel will illuminate the “Phantom” indicator if any of the input channels have Phantom Power enabled.
    • Phantom power is appropriate for use with condenser microphones or other devices that can (and must) be powered by the preamp. Mobile I/O limits the amount of phantom power to 10mA per channel, preventing device damage due to shorts. Some (rare) microphones may require more power than is provided by 10mA; you will need an external power supply to power those mics.
  3. Level Standard popup menu
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    Level Standard Pop-up Menu
    • This control allows you to select the input level and impedance characteristics for the input channel. The available choices are:
      • Line +4 — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, and the inline pad is engaged. Phantom power is not available. This format is appropriate for interfacing with professional audio equipment.
      • Line -10 — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, and the inline pad is engaged. Phantom power is not available. This format is appropriate for interfacing with prosumer and consumer audio equipment.
      • Inst (Instrument) — This format supports input levels up to +6 dBu. The input impedance is approximately 200k, and the inline pad is not engaged. Phantom power is not available. This format is appropriate for interfacing with high output impedance sources like guitar pickups.
      • Mic — This format supports input levels up to +6 dBu. The input impedance is approximately 200k (12k with phantom power engaged) and the inline pad is not engaged. Phantom power is available. This format is appropriate for interfacing with dynamic and condenser microphones on low to mid level sources. You may find that for very low level sources, especially with low-sensitivity microphones, that your SNR improves if you take advantage of a high-quality external mic preamp. External preamps are recommended for certain types of recording and microphones, especially classical recording and ribbon mics.
      • Mic/Pad — This format supports input levels up to +26 dBu. The input impedance is approximately 10k, (6k with phantom power engaged) , and the inline pad is not engaged. Phantom power is available. This format is appropriate for phantom powered condenser mics that put out near-line level signals (often the case when you are using condenser microphones on bass drums or other very loud sources).
  4. Gain Trim knob
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    Gain Trim Knob
    The Gain Trim knob allows you to adjust the analog gain of the input stage over a 40dB range determined by the input standard that you have selected. The gain is indicated in dB relative to the nominal level of the input standard you have selected. The gain changes are smoother near the bottom of the scale, with the steps increasing in size as you reach the +40dB gain limit.
  5. Channel Label
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    Channel Label
    This simply labels which channel is associated with the channel strip. Click in the lable to edit the channel name.
  6. Channel Level Meter
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    Channel Level Meter

    This is a peak reading, high-resolution, fast PPM meter. It shows the post converter level of the input signal of the associated channel.. The peak hold bar indicates the highest level seen on the channel since the last reset. You can reset the hold by clicking on the meter. These meters are simply high resolution versions of the meters shown on the front panel of the box – all the meter data is generated by the Mobile I/O hardware.

OPTIMIZING INPUT LEVELS

The Analog to Digital converters (ADC) in most devices function best when the peak level is around -6 dBFS (lowest distortion, best sound). This is true of the ADCs in Mobile I/O. Since you have full level control of the input with the gain trim knob, you will find that you get the best quality recordings if you try to set the nominal peak level of the input at about -6 dBFS. In addition to providing the best recording quality, it has the added benefit that you will be operating with an extra 6 dB of headroom before clipping. There is no drawback to optimizing your levels in this way, and plenty of benefit.

ANALOG INPUT CHANNEL LINK

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Analog Input Channel Link

In addition to the channel specific controls, each channel pair shares a Link button. When the Link button is engaged, changes made to one channel of the pair will automatically be applied to the other channel of the pair. This is very useful if you are miking with a stereo pair and need to maintain level balance between the two preamps -- with the Link button engaged, the balance is automatic and exact. When you enable the Link button, the control values for the odd channel in the pair will be copied to the even channel of the pair.

DIGITAL INPUT METERS

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Digital Input Meters

To the right of the Analog Input control section is the Digital Input Meter section. This group of meters provides level metering for all of the digital inputs on the Mobile I/O. These meters have the same response characteristics as the analog input meters and show you the audio activity on ADAT channels 1-8 and digital input channels 1-2, going from left to right.

Please note that when the box is clocking at 2x rates (88.2-96k), the ADAT input uses pairs of optical channels to transport the audio, and only channels 1-4 will show activity – channels 5-8 will not display any signal.

The label above the bank of 8 channel meters is a popup control that allows you to select between ADAT optical format and TOSLINK Optical SPDIF for the optical input connector. Click on the control to popup a menu to select between the different modes of operation. The mode is independently selectable for both the input and output.

If you have selected TOSLINK for either the input or output section, you will only see meters for channels 1+2 of the optical digital section. These correspond to the levels that are being transmitted or recieved in the optical SPDIF signal.

CANS CONTROLS

The next control block to the right is the Cans (Headphones) control section.

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Headphone Controls

The Headphone level control knob provides about 40dB of analog gain control on the headphone output. The Mute and Dim buttons reflect the state of the hardware mute and dim controls for the front panel headphone output. Engaging one of these buttons will also engage the associated control on the hardware.

When the Mute function is engaged, the headphone output will be muted. When the Dim function is engaged, the headphone output will be padded by 18 dB. The meters in this section show the headphone output pre-mute/dim block.

SYSTEM CONTROLS

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System Controls

The System block provides controls that adjust various system level aspects of the Mobile I/O hardware:

  1. The Clock Source popup menu controls the system clock source used by the hardware for digital synchronization and driving the converters:
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    Clock Source Pop-up Menu
    • Internal causes Mobile I/O to use its internal clock. You must select this if you want to set the sample rate from the Mobile I/O. If any other clock source has been selected, the console will not allow you to change the sample rate since the sample rate is determined by the external clock source.
    • WC (44/48) directs Mobile I/O to clock off of an external Word Clock Source at single rate (e.g. fs = 32k-50k)
    • WC (88/96) directs Mobile I/O to clock off of an external Word Clock Source at double rate (e.g. fs = 64k-100k)
    • WCx256 (44/48) directs Mobile I/O to clock off of an external 256fs Clock Source at single rate (e.g. fs = 32k-50k)
    • WCx256 (88/96) directs Mobile I/O to clock off of an external 256fs Clock Source at double rate (e.g. fs = 64k-100k)
    • ADAT (44/48) directs Mobile I/O to clock off of the incoming ADAT stream and run at single rate. All 8 ADAT channels are available. You will generally want to select this source if you intend to use ADAT input.
    • ADAT (88/96) directs Mobile I/O to clock off of the incoming ADAT stream and run at double rate. The bottom 4 ADAT channels are multiplexed over the lightpipe to provide 4 channels of double rate audio compatible with other SMUX and Alesis 96k ADAT devices. If you are using a device that provides 4 channels of 96k audio over ADAT optical, you will want to select this clock source.
    • DigIn (44/48) directs Mobile I/O to clock off of the selected stereo digital input at single rate (e.g. fs = 32k-50k). This allows operation of the digital input without SRC, and from devices that must supply clock.
    • DigIn (88/96) directs Mobile I/O to clock off of the selected stereo digital input at double rate (e.g. fs = 64k-100k). This allows operation of the digital input without SRC, and from devices that must supply clock.
  2. The Sample Rate popup menu allows you to select the sample rate when you are using internal clock. The Mobile I/O must be running on internal clock for the Sample Rate popup menu to have any effect. If the Mobile I/O is running from an external clock source, you cannot select the sample rate since it is determined by the external clock source.
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    Sample Rate Popup Menu
  3. The WC Out popup menu allows you to select the output clock signal the Mobile I/O generates on its WC Out BNC connector. The available choices are 1x and 256x. The 1x signal is appropriate for driving devices that accept a Word Clock signal. The 256x signal is appropriate for driving devices that accept 256x or SuperClock signals. Refer to the documentation for the external device to determine what is the most appropriate clock reference for it.
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    WC Out Popup
  4. The DI Source popup menu allows you to select the active input for the digital input pair. The choices are AES and S/PDIF. This selector physically switches the input to the digital audio receiver between the RCA input and the XLR input.
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    DI Source Popup
  5. The DI SRC button enables and disables the asynchronous sample rate converter (SRC) in the Mobile I/O digital audio receiver. When the SRC is engaged (button illuminated yellow), the digital audio receiver will automatically synchronize the input signal to the Mobile I/O system clock over a wide range of sample rate ratios.
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    DI SRC Buton
    This allows you to, for example, digitally transfer a sample from a CD player into a 96k session without any clocking problems. If you want to make bit-transparent transfers, you will need to disengage the SRC and ensure that the Mobile I/O and the external device are both using the same digital audio clock via one of the Mobile I/O synchronization mechanisms.
  6. The Lock indicators show which elements of the Mobile I/O clocking system are properly locked. The clocking system must be locked for the unit to behave as expected. If the system is not locked, audio will play at the wrong rate and will be distorted or noisy. Under normal circumstances, the system should always be locked, but if you have selected an external clock source and the clock signal is not present, corrupted or out-of-range, the system may unlock. There are indicators for the system and the digital input.
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    Lock Indicators

ANALOG OUTPUT CONTROL

The bottom half of the panel is dedicated to the hardware outputs of the Mobile I/O. Much like the analog inputs, each of the eight analog outputs has a set of controls associated with it. The controls are similar to the analog input controls. Each output channel has:

  1. Parameter Popup control — The Parameter Popup control allows you to save, recall, and manage all of the parameters associated with an analog output in one place. The control is documented in detail later in this chapter. All presets are automatically shared between all of the output channels.

  2. Level Standard popup menu — This control allows you to select the output level for the channel. The available choices are:

    • Line +4 This format supports output levels up to +26 dBu. The standard setting with the trim knob set to 0dB yields a +24dBu output when the digital signal driving the DAC is 0dBFS. This corresponds to +4 dBu nominal with 20dB of digital headroom. You can use the trim knob to adjust the analog output level to be consistent with the needs of other audio gear. The output impedance is approximately 50 Ohms. This format is appropriate for interfacing with professional audio equipment.
    • Line -10 This format supports a nominal output level of -10dBV with 20 dB of digital headroom. You can use the trim knob to adjust the analog output level to be consistent with the needs of other audio gear. The output impedance is approximately 50 Ohms. This format is appropriate for interfacing with prosumer and consumer audio equipment.

  3. Gain Trim knob — The Gain Trim knob allows you to adjust the analog gain of the output stage (post DAC) over a 40dB range determined by the output standard that you have selected. The gain is indicated in dB relative to the nominal level of the output standard you have selected. The gain changes are smoother near the bottom of the scale, with the steps increasing in size as you reach the +40dB gain limit. Unless the signal is quite low, you will put the output stage into analog clipping well before you hit the 40dB gain limit.

  4. Channel Label — This simply labels which channel is associated with the output.
  5. Channel Level Meter — This is a peak reading, high-resolution fast PPM meter. It shows the pre converter level of the output signal of the associated channel. The peak hold bar indicates the highest level seen on the channel since the last reset. You can reset the hold by clicking on the meter. These meters are simply high resolution versions of the meters shown on the front panel of the box – all the meter data is generated by the Mobile I/O hardware.

ANALOG OUTPUT CHANNEL LINK

In addition to the channel specific controls, each channel pair shares a Link button. When the Link button is engaged, changes made to one channel of the pair will automatically be applied to the other channel of the pair. This is very useful if you are driving a stereo monitor section or stereo device. By engaging the Link button, you will ensure that both channels have precisely the same amount of analog gain applied post DAC. When you enable the Link button, the control values for the odd channel in the pair will be copied to the even channel of the pair.

DIGITAL OUTPUT METERS

To the right of the Analog Output controls is the Digital Meters section. This group of meters provides level metering for all of the digital outputs on the Mobile I/O. These meters have the same response characteristics as the analog output meters, and show you the audio activity on ADAT output channels 1-8 and digital output channels 1-2, going from left to right. Please note that when the box is clocking at 2x rates (88.2-96k), the ADAT output uses pairs of optical channels to transport the audio and only channels 1-4 will show activity, channels 5-8 will not display signal. Both channels of the stereo digital output remain active at all sample rates.

The label above the bank of 8 channel meters is a popup control that allows you to select between ADAT optical format and TOSLINK Optical SPDIF for the optical output connector. Click on the control to popup a menu to select between the different modes of operation. The mode is independently selectable for both the input and output.

If you have selected TOSLINK for either the input or output section, you will only see meters for channels 1+2 of the optical digital section. These correspond to the levels that are being transmitted or recieved in the optical SPDIF signal.

BOX INFO

The Box Info section of the panel, in the lower right-hand corner of the window, shows you information about the currently connected and selected Mobile I/O unit. This section displays the Serial Number, Model Information and Firmware revision of the connected box. All of this information can be useful in trying to track down any connection problems that may arise.

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Box Info

If there is no information displayed in the Box Info section, the software is not communicating properly with the Mobile I/O hardware, or there is no Mobile I/O present on the FireWire bus.

If the FireWire light on the front panel of the Mobile I/O is illuminated but the box information does not appear in the console window, it is very likely that the software has not been installed properly. If this is the case, please refer to the installation instructions for details on how to properly install the software. Also, pleaseensure that there are no copies of the MobileIO Driver in the same folder as the MIO Console application or any of the ASIO drivers that you will use with an ASIO host.

If, on the other hand, the FireWire light on the front panel of the Mobile I/O is not illuminated, the box is not communicating properly with the computer. Please check the cabling of your Mobile I/O and other devices on the FireWire bus and make sure that everything is connected correctly. If that does not properly establish the connection, try rebooting your computer. As a last resort, try connecting only the Mobile I/O to the computer to ensure that communication can be established.

MIXER PANEL — For Legacy Boxes (without 2d Card)

The Analog I/O panel provides all of the bread-and-butter functionality that you would expect from a flexible audio interface like Mobile I/O - what you need to get the job done quickly and easily. The Mixer panel provides features that you won’t find anywhere else – here is where things start to get really interesting.

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Mixer Panel
The controls in the top half of the Mixer Panel are the same analog input controls described in the section I/O Panel. The bottom half of the mixer panel provides a control surface for the multiple integrated hardware mixers running on Mobile I/O. You access this panel by clicking on the “Mixer” button of the view panel selector bar at the top of the console window.

MIXER CONTROLS OVERVIEW

Each mixer has controls for Level (fader), Solo (button), and Mute (button) for each channel. Each stereo mixer also includes a pan control (knob) for each mono channel. For stereo inputs, such as the defaults of the digital inputs or your DAW output, will not have a pan knob. Instead, the two individual channels that make up the pair will be hard panned, and the fader meters will appear as stero meters.

At the far right of each mixer surface is the Master Fader. This fader will be Mono for Mono Mixers and Stereo for Stereo Mixers. Above the Master Fader is a bus mute button that allows you to quickly mute the entire mixer.

Each mix bus can have up to 36 inputs assigned to it. All of the analog, digital, and ADAT hardware inputs are available in each mixer. You can also assign any (or all) of the ASIO playback channels from your DAW to each mixer. Since there are 18 hardware inputs and 18 FireWire return busses that are routed from the ASIO driver on the host computer, you can mix up to 36 channels on each bus.

The default state of the hardware input channels in the mixers is faders at unity, mutes on, solos off and center panned. The playback channels from your DAW are unmuted in pairs per bus (e.g. DAW 1/2 will be unmuted on Mix 1/2, DAW 3/4 will be unmuted on Mix 3/4, etc.) MIXER PANE TABS

Use the tabs above the pan knobs to select the mixer you want to control. Each tab represents a mixer you have configured in the routing panel. When you click on a tab, the controls will be instantly updated to reflect the state of the associated mixer. In this way you can quickly switch back and forth between multiple independent mixes, each with a full mixing interface.

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Mixer Pane Tabs

Each mixer also has a Parameter Popup control associated with it. Unlike the Input and Output controls, the parameter control popup does not have an explicit user interface element representing it. Instead, the popup is accessed from the mixer’s tab itself. To pop-up the menu, you either click and hold the tab, or <control>-click the tab. The parameter popup menu allows you to maintain a library of standard mix configurations, scenes, and setups. It also provides a very quick method for copying mixes from one bus to another.

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Mixer Parameter Popup

CHANNEL FADERS

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Channel Fader

The Channel fader controls the relative level of the input channel in the mix. It works just like its non-virtual counterpart. The calibration numbers to the left of the fader knob provide an accurate guide to the amount of gain that will be applied by the fader. The label area above the fader knob lets you know which input channel the fader is controlling. The labels default to the console’s default names for the channels, but you can rename the channels to more meaningful names using the naming controls in the Routing Matrix.

The exact amount of gain for the fader is displayed in the small black window above the fader channel label. If you want to set a channel gain precisely, simply click in the small black window and type in the desired gain in dB numerically. Hitting the <return> key or clicking outside the entry box makes the new setting take effect. Hitting the <tab> key will make the new setting take effect and will move to the next numeric entry field in the mixer.

The gain range for each channel is from -∞ (muted) to +10 dB. The resolution of the gain control is extremely fine and gain changes are interpolated in the mixer, so there is no zipper noise when you make continuous (or even discontinuous) gain changes. <command> clicking the fader knob will allow you to make fine adjustments to the fader level by dragging the mouse up and down. <option> clicking the fader knob will reset the fader to unity gain. Clicking the meter associated with the fader will clear the peak holds for that channel.

For mono channels, the current gain is applied to the channel before the pan. For stereo channels, the current gain is applied uniformly to both of the hard-panned input channels.

CHANNEL METERS

Each channel fader has a meter (or a pair of meters for stereo channels) associated with it. The meters are calibrated consistently with the fader calibration. Each meter is a peak reading, fast-PPM meter that is pre-fader. The peak hold bar shows the biggest peak since the last reset. Clicking on a meter will clear the peak holds for that channel.

CHANNEL PANS

Each mono input in a stereo mixer has a pan knob above the fader. The pan knob allows you to control the relative amount of the input channel that is placed into the two busses of the stereo mixer. Panning hard left (L100) means that the channel will appear at full volume in the left (odd) bus of the mixer. Panning hard right (R100) means that the channel will appear at full volume in the right (even) bus of the mixer. When the channel is center panned, the signal appears at a decreased volume (-3dB) in both channels, such that the volume of the total signal in both channels is equivalent to full volume in one channel. As you pan from left to right, the signal is distributed between the two channels so that the total volume remains constant.

As with the channel fader, the black window above the channel pan knob provides a precise readout of the current pan position, and clicking on the black window will allow you to type in an exact pan amount. Negative numbers (-100 to 0) indicate left pans and positive numbers (0 to 100) indicate right pans. 0 is center pan. <option> clicking the pan knob will return it to center pan, and <command> click-dragging the knob will allow you to adjust the pan position in finer increments.

MUTE BUTTONS

Each input has a Mute button (labeled “M”) associated with it. When the mute button is engaged (illuminated white), the channel will be muted in the mixer. The mute is interpolated, so muting a channel will not cause audible clicks. <option>-clicking a mute button will set all of the mutes on the mix bus to the same state as the button you click. This allows you to quickly mute all the channels or quickly unmute all the channels on a bus.

SOLO BUTTONS

Each input has a Solo button (labeled “S”) associated with it. When the solo button is engaged (illuminated red), only channels that have been soloed will be mixed by the mixer. The other channels will effectively be muted. As with the mute button, the gain changes associated with soloing or unsoloing a channel are fully interpolated and will not cause audible clicks. <command>-clicking a solo button will exclusively solo the associated channel. The <command>-click will automatically clear the solo state of all other channels on the mix bus. <option>-clicking a solo button will set all of the solos on the mix bus to the same state as the button you click. This allows you to quickly solo all the channels or quickly un-solo all the channels on a bus.

MIXER MASTER FADER

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Mixer Master Fader

The Fader that appears on the right side of the mixer pane is the Mix Master Fader. This fader controls the overall bus level of the mix bus. Operationally, it is exactly the same as the channel faders (see: Channel F). Each Mix bus in the system has 24 dB of headroom above full scale. If the summing point of the mix bus is clipping, you can pull the mix out of clipping by dropping the Master fader, as long as the sum point is clipping by less than 24 dB.

MIXER MASTER MUTE

Each mix bus has a master mute button above the master fader that allows you to mute the output of that mixer. This mute is interpolated.

WIDE MIXERS

As was described before, the integrated mixers in Mobile I/O are WIDE – that is they allow you to mix every available input channel (both hardware channels and FireWire channels) together. In the widest case, the mixer will have 36 faders to allow you to control the gains for all of the input channels.

The MIO console window is only wide enough to accommodate 18 faders plus the Master fader. If you have enabled more channels than will fit in the window, the MIO console will automatically display a scrollbar at the bottom of the Mixer pane. Use this scrollbar to control which faders are visible at any given time. The width of the scrollbar indicator shows you how many of the enabled channels are visible at any given time. The scrollbar will automatically be hidden if you reduce the number of inputs to the mixer below 19 channels or if you switch to a different mixer that has less than 19 channels enabled.

TIP: Mobile I/O’s support of near-zero latency mixing of every channel opens up a huge variety of applications that cannot be achieved with standard interfaces or, at the very least, require external gear or major work-arounds to accomplish. Some examples are:

  • Stem-based mixing.
  • In this mixing technique, you mix disparate elements of the program to separate sub-mixes called stems. You might mix drums to one stem, instruments to another, and vocals to a third. Then the relative balances of the mix can be addressed later in a macroscopic way (during mastering, for example). This also enables remixing the project easily without having to go back to the multi-track master. Since you will be creating stems on individual busses in the DAW, you need to sum the stems for monitoring. This is easily accomplished with Mobile I/O’s mixer.
  • Monitoring mixed-in effects sends when the external effects are unavailable.
  • This is Mobile I/O, right? You may find that you want to continue editing or mixing while you are away from the studio. You can use the WIDE mixer in Mobile I/O to mix in DAW effects send busses for monitoring without having to reconfigure your DAW session.
  • Multichannel foldback mixes.
  • Ever tried monitoring a full drum kit with a host-based foldback mix, or even a 2 channel cue mix? It can’t be done! At least, it can’t be done well). To monitor multichannel input tracking sessions you need an external mixer – or the built in WIDE-mixer in Mobile I/O.
  • If you take advantage of MIO’s ADAT input with a third party A/D converter (or a second MIO) with ADAT outputs, you can simultaneously track and mix up to 18 input channels, with no latency, and no external mixer!
  • Near-Zero Latency monitoring of external effects.
  • Most singers need some reverb or other effects to get the feel right during their performance. With Mobile I/O and the WIDE mixer, you can split off a send from the performer’s input channel, send it to an external effects unit, and mix the effect return into the performer’s foldback mix -- with virtually no latency.
  • Multiple WIDE mixes
  • Since Mobile I/O supports multiple WIDE mix busses simultaneously, you can form multiple, individual foldback mixes for multiple performers at the same time, and best of all, each mix has its own complete mixer control surface, so you don’t have to mess around with a million unreadable aux send knobs.

ROUTING PANEL

The third panel in the MIO Console window is the Routing panel. This panel is what you use to access the powerful routing features that you won’t find anywhere else. The Routing Matrix portion of the panel lets you dynamically control the configuration of the WIDE mixing engine and also configure the channel names for the hardware. You access this panel by clicking on the “Routing” button of the view panel selector bar at the top of the console window.

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Routing Panel

The Routing Panel has two main components:

  1. The Output Patchbay on the left side of the panel.
  2. The Routing Matrix on the right side of the panel.

THE ROUTING MODEL

The Routing Panel provides a user interface for controlling the underlying routing architecture of the Mobile I/O. The routing architecture provides a powerful routing model to allow you to control the routing of signals between physical & virtual inputs and the hardware mixer & physical outputs.

Conceptually, the architecture is quite simple:

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Block diagram of the Mobile I/O routing architecture

All of the physical inputs (e.g. Analog, ADAT and Digital), all of the channels being transmitted over the FireWire bus from the ASIO Application (e.g. DAW), and all of the outputs from the Mobile I/O WIDE Mixer are available to the Output Patchbay. The Output Patchbay can cross-point assign any of its inputs to any of the physical outputs (e.g. Analog, ADAT and Digital).

All of the physical inputs and all of the channels being transmitted over the FireWire bus from the ASIO Application are also inputs to the Mobile I/O WIDE Mixer. Every bus has each of those inputs available for mixing. The mixer outputs are sent to the Output Patchbay for routing to physical outputs. The number of mix busses varies with the Mobile I/O hardware model and the sample rate. The Mobile I/O 2882 supports 10 mono (5 stereo) mix busses at sample rates up to 50kHz (e.g. 1x rates) and 4 mono (2 stereo) mix busses at sample rates up to 96kHz (e.g. 2x rates).

Every physical input is mult’ed from the router/mixer section and sent directly over the FireWire bus to the ASIO host. Regardless of any mixing, routing, or mult’ing that you configure in the hardware, you can always record all of the inputs with your DAW.

As you can see from this simple, high level view, the Mobile I/O routing architecture supports direct routing of any input to any output and also mixing of any set of inputs to multiple mixers. The outputs of the mixers can be routed to any output or any set of outputs for hardware mults. All hardware inputs are available to the ASIO host.

As it turns out, the Mixer/Router in Mobile I/O is quite extensive and the simple user interface hides a lot of complexity:

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Complete view of the Mobile I/O Matrix Mixer

As you can see from this schematic diagram of the mixer/router, the Mixer in Mobile I/O is a fully cross-pointed matrix. While we are presenting the mixer to you as a set of mono and stereo mixers, the actual underlying structure is a full matrix. This means that the mixer is ready for surround and multi-channel matrix mixing today. The schematic provides an accurate representation that shows the signal flow through the mixer and into the router with all analog and digital gain points represented.

OUTPUT PATCHBAY DETAILS

The Output Patchbay allows you choose the signal sources to send to the hardware outputs. You can feed any output from your choice of a mix bus, a hardware input, or a FireWire input from the computer.

PATCHBAY POPUP CONTROLS

The output patchbay is the set of popup menus on the left side of the panel. Each popup menu corresponds to the hardware output that is labelling it. The channel that you choose from the popup menu will drive the associated hardware output.

Since the output patchbay is fully cross-pointed, you can easily construct direct routes from hardware inputs to hardware outputs for stand-alone converter operation, routes from ASIO (using the DAW channels) to the outputs for direct dubbing and monitoring from the computer, and routes from the mixer to hardware outputs for foldback, monitoring, and effects sends.

The output patchbay even lets you send the same channel to multiple outputs, which makes it really simple to create channel mults. You can use this to:

To choose a source for an output channel, simply go to the popup menu for the output you want to route to, click and select the source send. Each popup window is like the jack of a patchbay, and each name in the popup menu is like a patchbay cable that carries the named signal.

PATCHBAY PARAMETER POPUP CONTROL

Directly above the patchbay popup menus is the “patchbay parameters” parameter popup control. This control allows you to store patchbay configurations for later recall. See the section patchbay popup controls, for more details about the parameter popup menu.

The factory default routing for Mobile I/O configures the box to behave as a direct-routed audio interface – all the inputs go directly to the computer, all the outputs from the computer are directly routed to the hardware outputs, and the Headphone output is fed by the stereo mixer on Mix 1/2. This makes the Mobile I/O act like a basic audio interface.

Other configurations that may be very useful include:

This control allows you to maintain a library of frequently used routings and switch between them at will. Spend a little time to familiarize yourself with it — it is extremely powerful.

ROUTING MATRIX DETAILS

The Routing Matrix (Matrix) is the mixer assign matrix that is on the right side of the pane (see “Routing Panel” ). This control surface allows you to fully configure the structure of the multi-bus WIDE mixer in the Mobile I/O hardware. It also allows you to name all of the hardware and virtual channels that are accessible in the hardware.

You use the Matrix to assign hardware and FireWire inputs (DAW outputs) to mixers in the Mobile I/O. To assign channels to a given mix, you click on a crosspoint in the Matrix, darkening the associated Crosspoint Assign tile. Deselecting a tile removes the associated fader from the mixer and mutes the channel in the hardware.

The WIDE mixer allows you to assign all of the hardware input channels and all of the FireWire channels to each mix bus. By limiting the number of channels assigned to any given bus, you can reduce the complexity of the associated mixer interface.

LOGICAL DESCRIPTION

Each mix bus and each input channel has a path label tile associated with it. These tiles are arrayed along the top and left edges of the Matrix.

The tiles along the top edge of the Matrix are the input path tiles. Each tile is color coded based upon the type of input it is:

Each tile shows the Mobile I/O unit it is associated with, the physical name for the path, and the user defined name for the path.

Above each pair of input tiles is a thin tile that is used to join two input paths into a stereo input channel. In the factory default configuration, the Digital Inputs are joined into a stereo pair and each pair of DAW channels is joined into stereo pairs. If the channels are joined into a stereo pair and the pair is assigned to a stereo mixer, the pair will be represented by a single stereo fader in the mixer interface, and no pan knob will appear.

The white tiles along the left edge of the Matrix are the Mix bus tiles. All of these tiles are white because they all represent mix busses. Each tile shows the Mobile I/O unit it is associated with, the physical name for the mix bus, and the user name for the mix bus.

To the left of each pair of Mix tiles are thin inset tiles that are used to join two mix busses into a stereo bus. If the busses are joined, the associated mixer will have pan knobs for each of the mono hardware channels. If the bus is a mono bus, there are no pan knobs associated with the input channels.

The interior of the Matrix is composed of a large number of square crosspoint assign tiles. The Matrix is too wide to fit completely within the MIO Console window; you may use the scrollbar that appears at the bottom of the Matrix to scroll the remaining DAW channels into view.

Each crosspoint assign tile indicates whether or not the input channel at the top of the column is assigned to the mix bus at the edge of the row. If the tile is filled in, the channel is assigned to the mix, and the controls for that channel will appear in the associated mixer. If the crosspoint tile is not filled in, the channel is not assigned to the mix, and the controls will not appear. The channel will be muted and un-soloed on the associated mix bus.

Stereo pairs are automatically assigned as a group to busses.

There are some keyboard shortcuts that you can use when making Matrix assignments:

CONFIGURING CHANNEL NAMES IN THE MATRIX

MIO Console has fully user configurable channel names. The names that you select for your channels will propagate to all of the other aspects of the MIO Console user interface.

This allows you to name the channels in meaningful ways. The analog inputs can be named to match the sources. The Digital I/O can be named to match the effects device that you have patched. Mixes can be named by the foldback monitor or effects send that they will feed.

To name a channel, click on the input path tiles (for input channels) or the mix bus tile (for Mix busses). The channel configuration window will appear above the MIO console window:

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Channel Configuration Window

All of the channel identification controls will be set for the tile you clicked on. To change the user selectable name of the channel, simply type the new name. The name will be updated in the console when you do one of the following things:

You can dismiss the window without updating the channel name by clicking its close box.

You can also select the stereo linking state from Linking popup menu in the channel configuration window. The state will be updated in the console along with the channel name.

The channel identification controls identify which channel you are adjusting:

MATRIX PARAMETER POPUP

The entire state of the matrix can be saved to and recalled from the console parameter library system. The basic functions of the parameter popup control are documented in the next section. Since you are very like to have a number of tracking and mixing configurations you use over and over again, the Parameter Popup for the Matrix is a real timesaver. This control appears in the top left corner of the pane, above the Parameter Popup control for the Patchbay router. Each time you create a configuration that you are likely to use again, save it in the Parameter Library for instant recall when you need it next.

PARAMETER POPUP CONTROLS

The Parameter Popup control is MIO Console’s unified mechanism for handling presets for the various sections of the Mobile I/O. Each element of the console that supports the Parameter Library mechanism has a parameter popup control associated with it. These elements currently include:

Each instance of the Parameter Popup control provides the same commands and options for every section of the console.

POPUP COMMANDS

The parameter popup provides a hierarchical, categorized library of configuration presets for the associated section of the console. The menu is divided into three portions. The first portion consists of all of the items above the “Factory Default” item. The second portion is the “Factory Default” item and the third portion is the hierarchical items below the “Factory Default” item (see Parameter Popup Menu ).

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Parameter Popup Menu

The commands in the first portion of the menu allow you to save and manage the presets in the library. All of the presets are shared between like elements in the console. The preset commands are:

The “Factory Default” command will set the current state of the associated console settings to the default settings.

POPUP PRESETS

In the third part of the menu, each of the categories will be listed as a hierarchical menu title. Each of the presets for each category will be listed in the submenu under the category menu. The currently selected category and preset are drawn in bold, so you will know what is currently active.

Selecting a preset from the menu will make that preset active and will set the current state of the associated console settings to the values contained within the preset. The name of the currently selected preset will be drawn in the popup area in the console window to indicate which preset is active.

If you change the settings in the console, the name of the preset will be drawn in italics indicating that the current settings differ from the selected preset.

For the Input and Output channels, you can hold down the <option> key while selecting a preset to automatically apply the preset to all of the other input or output channels.

To access the parameter popup for the mixers, either click and hold the associated mixer tab or <control> click the associated mixer tab.

We have provided an initial set of presets for the various parameter libraries. The presets for the output channels are relatively complete and give you an idea of the power and flexibility of this approach to parameter management. We will be adding presets on a regular basis – check the website for new presets.

PERSISTENT STATE MANAGEMENT

All Mobile I/O hardware has support for setting a Boot State — the configuration the hardware will use when the unit boots up. As of v.5 of the Mobile I/O software, this boot state includes the entire state of the unit including the configuration of the mixer, the router, sample rate, clocking analog I/O levels (for HW that has digital control), and +DSP configuration.

This functionality allows you to fully configure your hardware and “pour” a complete digital signal processing engine into the HW for instant-on processing.

To configure the Boot State for your Mobile I/O:

  1. First, attach the Mobile I/O to the computer and start up MIO Console.
  2. Use MIO Console to configure the box. Set up all aspects that you care about. Once you have the configuration as you like it, you are ready to save the snapshot.
  3. Choose the “Save Boot State” command from the “Utilities” Menu

The ULN-2 hardware extends the Boot State and adds support for Persistent State Snapshots. There are 10 snapshot slots in the ULN-2 that are recallable from the controls on the ULN-2 front panel. Each Persistent State Snapshot contains a complete description of the state of the box, including Sample Rate, Clock Source, Digital input source, Sample Rate Converter Enable, Patchbay routing, Mixer Configuration, Levels and +DSP configuration and routing. In other words, a snapshot saves every aspect of the configuration of the ULN-2.

The first snapshot slot is special as it is used by the unit to configure the hardware and the routing when the ULN-2 starts up. The other 9 slots are available for storing alternate configurations that can be selected “on the fly” after the ULN-2 is up and running.

When a computer is attached to the ULN-2, the front-panel controls to select snapshots are locked-out since the computer is actively controlling the configuration of the box.

If the computer is not attached, the two tact-switches on the left-side of the front-panel (between the status indicators and the meters) may be used to select the snapshot that you want to use to configure the ULN-2. These buttons are labled with up and down arrows. The currently selected snapshot is indicated by the column of LED’s labled C, 1, 2, 3, 4, 5, 6, 7, 8, 9. When the ULN-2 turns on, the “C” indicator will be illuminated, indicating that the unit has booted up with the state that was stored in the “Boot Snapshot”.

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ULN-2 Front Panel Snapshot Controls

Pressing the up arrow will move to the next higher snapshot in the list (e.g. if you are currently on snapshot 3, you will move to snapshot 2). Conversely, pressing the down arrow will move to the next lower snapshot in the list (e.g. if you are currently on snapshot 3, you will move to snapshot 4). If you are at either beginning of the list and you press the up arrow, you will wrap around to the last item in the list. When you select a new snapshot, the new snapshot is applied to the box immediately.

In order to configure the boot state and snapshots for your ULN-2, you will need to utilize the MIO Console application. Configuring and storing snapshots in the box is very simple:

  1. First, attach the ULN-2 to the computer and start up MIO Console.
  2. Use MIO Console to configure the box. Set up all aspects that you care about. Once you have the configuration as you like it, you are ready to save the snapshot.
  3. Choose the appropriate save command from the Utilities Menu
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    Utilities Menu
    • To save the snapshot to the “Boot State” slot, choose the “Save Boot State…” item.
    • To save the snapshot to one of the other snapshot slots, choose the appropriate “Save Snapshot x State…” item (where x is the appropriate number).
  4. Save a copy of the current Console state to a file on your hard disk with an appropriate name (like “ULN-2 Snapshot 1” for the 1st snapshot) so that you have a copy of the state on the computer if you want to modify it in the future.

With v.5, the 2882 supports an alternate boot state as well. This is the state that is saved in the “Snapshot 1” state slot. If there in nothing saved in this slot, the alternate boot state will be the factory default boot state.

To access the alternate boot-state on the 2882, simply hold the front-panel DIM button while powering the unit. This will select the alternate boot state.

 MIOConsole Preferences

MIO Console Preferences

MIO Console has a number of preferences that you can set to control aspects of its behavior. These preferences are accessed via the MIO Console > Preferences… command (or via the <command>-, key sequence). When you select the Preferences command, the Preferences sheet is shown on the MIO Console window:

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MIO Console Preferences

The preferences you can control are:

 Record Panel

Record Panel

MIO Console integrates a dedicated multichannel recording interface. The Record Panel allows you to record with Mobile I/O right out of the box without needing to add any additional software to your system.

The Record Panel was purpose built for doing massive multichannel recordings with multiple boxes. We have deployed systems recording 72 channels at 96k, and have tested systems with even more channel capability. The Record Panel is not an editor; it does not support overdubs; it is optimized for capturing audio to disk with no muss and no fuss, with extreme reliability.

The Record Panel is accessed using the Recording panel button in the MIO Console window:

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Record Panel

Record Panel Description

The Record Panel UI has a number of elements that work together to allow you to configure and monitor your recording.

The first element is the time readouts:

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The Clock readout shows the wall-clock time and may be used for logging. It is formatted as HH:MM:SS.

The Record Time readout shows how long the recording has been running for when the Record Panel is in record mode, and how long the Record Panel has been playing for when in playback mode. It is formatted as HH:MM:SS.sss.

The Disk Time Remaining readout shows the time available on the selected recording disk. It automatically adjusts based upon sample rate, number of armed tracks, sample size and the amount of space available on the target disk. It updates as the recording progresses and is formatted as HH:MM:SS.

The next elements are the transport controls:

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The first button is the Stop button. Clicking this button will stop both the record and playback transport. When all transports are stopped, this button will be lit yellow.

The second button is the Play button. Clicking this button will start the playback transport if the RP is not already playing audio. If the RP is already playing, clicking the button will stop the playback transport. When the playback transport is running, this button will be lit green.

The third button is the Record button. Clicking this button will start the recording transport if the RP is not already recording audio. If the RP is already recording, clicking the button will terminate the current take and start the next one, continuing recording. When the playback transport is running, this button will be lit red (as shown in the picture above).

The next elements are the progress meters:

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The CPU Meter shows the amount of the real-time timeslice that is being consumed by the Record Panel.

The Rec Meter shows the amount of the record buffer that has been consumed by audio that has not been saved to disk. Under normal operation, this meter will pulse upwards and return to a value near zero periodically. If your disk is too slow, this meter will increase towards 100%, and if it reaches 100% it means that the disk is not recording fast enough and there will be glitches in the recorded audio. This meter is only active during recording.

The Play Meter shows the amount of the play buffer that is empty (not yet read from disk). Under normal operation, this meter will pulse upwards and return to a value near zero periodically. The RP uses an adaptive algorithm for filling the play buffer so the behavior of this meter may appear somewhat erratic. If your disk is too slow, this meter will increase towards 100%, and if it reaches 100% it means that the disk is not supplying data fast enough and there will be glitches in the playback audio. This meter is only active during playback.

The next element is the Play position meter:

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When audio is playing back, this meter indicates the current playback position in the take. You can click in this meter during playback to cue playback to a different location within the take.

The next element is the tracks overview:

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The tracks overview shows all the tracks available for recording. Tracks are added to the tracks overview in one of two ways:

For 2d Expanded boxes, the tracks appear in the same order that the FireWire assigns appear in the mixer, so you can choose the track order by reordering the mixer strips in the 2d Mixer UI.

For legacy boxes, the tracks appear in the same order that they do in the Matrix and as inputs to the computer.

So, you have much more control over what appears in the RP when you have a 2d Expanded box. In addition to the rules listed above for how tracks are added to the the tracks overview, there is a global preference (controlled via the Recording Menu) that allows you to control whether tracks from offline boxes (that obviously cannot be recorded from) should be listed in the tracks overview. It may be useful to list tracks from offline boxes if you are doing offline configuration, but in general you will not want to show tracks from offline boxes.

When you are recording, each track shows a continuously updating track overview for the signal that has been recorded for that channel. You can scroll back and forth in time; if the horizontal scroller is set to the far right, the track overview will autoscroll, keeping the updating area of the track overviews in view.

Each track has a track header:

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The track header shows the box, the FireWire channel and the channel name for the track. If the height of the tracks is too small for both lines to fit, only the channel name for the track is displayed. The channel name is automatically set by the source for the track; you can change the names by editing the input names appropriately.

To the immediate right of the label is a track record enable button. When the button is red, the track is record enabled and will be recorded the next time you hit record. While you can change the record enable state of the track while recording, it will have no effect on the current take — just the next take.

You can control the track enables directly from the Record Panel, or, if you are using a 2d Expanded box, you can control the record enables from the mixer. The mixer interface is especially convienent because one Record Enable button may actually control multiple tracks.

Finally, there are zoom buttons in the lower right corner of the window that control the zoom level of the tracks overview:

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The buttons with the + in the magnifying glass icon zoom in (increase magnification), and the buttons with the - in the magnifying glass icon zoom out (decrease magnification). The buttons stacked on the right side control the vertical zoom and the buttons along the bottom control the horizontal zoom.

Recording

When you click the Record button (or initiate record using a control surface or command key), the Record Panel recording engine creates a new Take Folder for you. The Take Folder is created in the Record Folder that you set using the Recording > Set Record Folder… menu command. If you don’t set a Record Folder before you start recording, the RP will default to using the Documents folder in your home directory.

Each time you start recording a new Take Folder is created. Depending on the state of the Name Take Folders incrementally recording preference, this Take Folder will be named in one of two ways:

Each Take Folder will contain one audio file for each track that is Record Enabled when you start the take. Each audio file will be named as follows:

TT-<trackname>.<ext>

Where:

The Take Folder will also contain a log file written by the recording engine.

While the RP is recording, each enabled track will show a continuously updating track overview that shows the history of the peak envelope of the channel. The the track overview will autoscroll as time progresses. You can use the scrollbars to adjust the currently displayed portion of the audio history, and the zoom buttons to control what period of time is displayed.

As each take is started, the current playback folder is set to the current take folder.

Clicking the Record button while the RP is recording will immediately start a new take.

Clicking Stop will end the current take (and stop any current playback as well).

Playback

The RP can playback take folders (actually any folder of audio files). The file types and bit depths of the files in the folder can be mixed (the playback engine will automatically adjust). The playback engine ignores the sample rates of the files in the take folder — it simply plays out the samples at the current sample rate of the hardware.

As described in the previous section, the RP automatically sets the current playback folder to last take folder (so you can play back the last take by just clicking play). If you wish to play back a different take folder, use the Recording > Set Playback Folder… menu command to choose the Take Folder you want to play back.

When you click the play button, the RP playback engine will find the size of the recording, load the audio into the playback buffer, and begin streaming it to the HW. The audio is streamed out on successive DAW channels starting at DAW 01 on the first box and increasing until the first box is full (18 channels for current HW), then moving on to DAW 01 for the next box if there are more than 18 channels. It continues on in this way until it runs out of audio files or boxes.

Files recorded by the RP and played back without any naming changes will be played back in track order. If the take folder does not use RP naming conventions, then the playback track order will be determined by the alphabetical sort-order of the audio file names.

You can cue within the take by clicking in the Pos meter. If there is no Recording active, the Record Time read out will show the current time of the playback position within the take. Once the playback engine has run out of samples on disk, the playback engine will stop.

If you want to stop playback without stopping a current record take, you can click the Play button again to stop playback.

The RP playback engine enables a couple of cool tricks:

Clicking Stop will end the current take playback (and stop any current recording as well).

Multibox considerations

The Record Panel is capable of recording from multiple boxes simultaneously up to very large track counts. The boxes can even be spread across multiple Firewire busses to support more boxes than can be transported on one Firewire bus. It is critical, however, that all the boxes in the system are on the same clock.

If you run the boxes wild, it may appear that the recording is functioning, but eventually the recording buffer will desynchronize and distortion or glitches will be introduced into the recording.

You can use any available clock source to ensure that the boxes are all on the same clock reference. Both AES and Word Clock are good choices.

The Record Panel will only record from a box if there are tracks enabled on that box. You can use this feature to chain one box to another via ADAT (for example) and effectively use one box as an expander for another. The expander box can be on the FireWire bus and controlled by MIO Console, but it will not use FW bandwidth or isochronous resources if there are no tracks enabled on it. This allows you to effectively double the number of boxes that can be added to one FireWire bus.

Record Panel Key Commands

MIO Console defines a number of key commands that you can use to control the Record Panel from the keyboard or a configurable HID device:

Record Panel Key Commands

Command Key Sequence
Record Panel: Zoom In Channels ⌘⇡ (Command + ⇡)
Record Panel: Zoom Out Channels ⌘⇣ (Command + ⇣)
Record Panel: Zoom In Timeline ⌘⇠ (Command + ⇠)
Record Panel: Zoom Out Timeline ⌘⇢ (Command + ⇢)
Record Panel: Scroll Channels Up ⇧⇡ (Shift + ⇡)
Record Panel: Scroll Channels Down ⇧⇣ (Shift + ⇣)
Record Panel: Scroll Timeline Right ⇧⇠ (Shift + ⇠)
Record Panel: Scroll Timeline Left ⇧⇢ (Shift + ⇢)
Record Panel: Play ⌘J (Command + J)
Record Panel: Stop ⌘K (Command + K)
Record Panel: Record ⌘L (Command + L)

You can change each of these key commands if you like. See MIO Console Key Commands for more details.

Record Panel Prefs

You control various aspects of the way that the Record Panel records using the Recording Preferences sheet. Access the sheet using the Recording > Recording Preferences… menu command or the Record Panel prefs button:

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The Recording Preferences sheet will appear:
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Recording Preferences sheet

You can specify various text strings that may be included in the metadata included with some files using the 4 text entry fields.

You may specify the file format that the Record Panel uses to record with the Record File Format popup menu which has the following choices:

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Record File Format Popup
You can choose the file format to use based upon the following criteria:

You can specify the bit depth that you would like to record at. Using 16-bit files saves space but provides less dynamic range. The Record Panel does not dither the incoming 24-bit signal to 16-bits (it is just truncated). You can use dither in the MIO Mixer to dither to 16-bit for recording if you will record at 16-bit.

If you record using the BWF file format, the Record Panel will include a BEXT chunk with recording metadata. The BWF Metadata fields allow the you to set the Country Code and Facility Code in the BEXT chunk. This probably is only useful to you if you work with an EBU facility.

Finally, there is a preference to control how Take Folders are named. Every time you hit record, the Record Panel will create a new Take Folder that will contain all the audio files for the take. If this checkbox is not checked, the Take Folder is named with a Date/Time stamp. If the checkbox is checked, each take will be sequentially named so that you will have a sequence of takes starting with 1 and incrementing by one each time you click record.

v.5 Monitor Controller

Monitor Controller

The Monitor Control window consolidates the most important monitoring functions for your MIO into one convenient window. You can configure which sources you want to monitor and from which outputs they will be monitored, for any number of channels from mono to 7.1 Surround. You can configure the Monitor Controller as a floating window so it always remains accessible, even when the MIO Console is hidden. You can choose between the full size Monitor Control window and a smaller Mini Controller window that contains nearly all the functions of its big brother. Finally, as an optional component of the MIO Console, if the Monitor Controller simply doesn’t fit into your workflow, you don’t need to use it.

Ultra-Quick Start Guide to Configuring the Monitor Controller

If you just want to set your main outputs of your Mix Bus to be controlled by the monitor controller, follow the steps below:

To Add the Bus Output of your Mix Bus to the Monitor Controller:
  1. In the Master strip for your Mix Bus in the v.5 mixer, click the Bus Output pop-up menu.
  2. Select Add to Monitor Controller. The Bus Output of your Mix Bus is now automatically configured as the Monitor Source for the Monitor Controller.
To Configure the Monitor Controller:
  1. Select Window > Show Monitor Controller.
  2. Click the Configure button at the bottom of the Monitor Controller window. The Configure Monitor Controller sheet appears:


  3. Click the ‘+’ button in the Monitor Paths section at the bottom of the Configure Monitor Controller sheet. The Add Monitor Output dialog appears.
  4. Enter the name of the new Monitor Output Path.
  5. Select the Bus Type of the new Monitor Source.
  6. Click OK. The new source will appear in the Monitor Source List.
  7. Click the pop-up menu in the “Signal Source” column for the Left Channel of the source. Select the appropriate physical source channel from the list. This includes Physical Input and DAW channels.
  8. Repeat step 7 for each of the channels that make up the bus.
  9. Click Ok. The Monitor Controller now has your Bus Output configured as its Monitor Source, and the Output Path configured above.

 

Monitor Control Interface and Basic Operation

The Monitor Control window interface has four sections:


In order to use the Monitor Controller, you’ll need to do the following:

  1. Configure one or more Monitor Sources
  2. Configure one or more Monitor Output Paths
  3. Select a Monitor Source to monitor
  4. Select an Output Path for your monitored signal

After completing these tasks, you can use the Monitor Control window to control all the monitoring functions for the selected Monitor Source, which you will hear via the selected Output Path.

MIO Console automatically saves your Monitor Sources and Output Paths, so you will only need to define each Monitor Source and Output Path once. You can use the Monitor Controller with only one Monitor Source and one Output Path defined, or you can define as many as you’d like of either or both and use the Monitor Controller to switch between them.

Determine Configuration

The most important step in working with the Monitor controller is to determine how signal is routed through your mixer. Once you know how your signal flows through the mixer to the Mix Bus, configuring and using the Monitor Controller is very simple. In order to configure the Monitor Controller you will need to identify two different types of audio channels in your system:

Identifying Monitor Sources

The selectable input source that you wish to monitor is called the Monitor Source. Your sources can come from the physical hardware inputs of the Mobile I/O or they can be virtual inputs streaming into the MIO over the FireWire cable, and can consist of any number of channels from a mono channel to eight channels forming a 7.1 surround signal. You can freely name your Monitor Source in order to more clearly identify what sources are being monitored. For example, if your monitor source was your AES inputs, you might want to name your Monitor Source “AES input” or simply “AES”. On the other hand, if you have a DAT player connected to your AES input, you might instead want to name the Monitor Source name “DAT”.

You can also use mix busses as sources in the Monitor Controller. To add a mix bus to the Monitor Controller, you simply select the "Add to Monitor Controller" item in the Bus Output pop-up menu at the bottom of the master strip for the bus. This automatically creates a Monitor Controller source for you. The mixer will automatically maintain the routing of the bus output to the monitor controller for you.

When you create a Monitor Source, you specify the type of path you are making, and the name of the new path. The type of path (Mono, Stereo, LCR, LCRS, Quad, 5.0, 5.1, 7.1) you choose will allow the Monitor Controller to automatically create channel slots for each component channel.

The Monitor controller uses the channel assignments to match the sub-channels between Monitor Sources and Monitor Output Paths and automatically route your selected source to your selected Monitor Output Path.

Once you have identified the sources you want to select from in your system, you use the Monitor Controller’s Configuration Dialog to set up the sources. See “Configuring Monitor Sources” for more details.

Identifying Monitor Output Paths

A Monitor Output Path defines a selectable destination for your monitor system. It can be anything from a single Mono channel, to a 7.1 surround path. Monitor Output Paths are associated with real physical outputs that you have connected to Monitoring devices (like Speakers/Amplifiers or Headphones). These destinations are unlikely to change frequently, but you may have more than one destination that you use routinely. For example, you may have the following monitoring systems in your studio:

  1. Console (or desk) mounted nearfield (small) stereo monitors.
  2. A larger far-field 5.1 surround monitor system.
  3. A pair of studio headphones.

You may want to monitor any of the Monitor Sources through any of these monitoring systems, but only through one at a time. So in this case you would configure three Monitor Output Paths, each with its own physical outputs — one for each system listed above.

As with the Monitor Sources, you choose the names for these Monitor Output Paths in a way that makes sense to you.

When you create a Monitor Output Path, you specify the type of path you are making, and the name of the new path. The type of path (Mono, Stereo, LCR, LCRS, Quad, 5.0, 5.1, 7.1) you choose will allow the Monitor Controller to automatically create channel slots for each component channel.

The Monitor controller uses the channel assignments to match the sub-channels between Monitor Sources and Monitor Output Paths and automatically route your selected source to your selected Monitor Output Path

Once you have identified the output paths you want to select from in your system, you use the Monitor Controller’s Configuration Dialog to set up the outputs. See “Configuring Monitor Output Paths” for more details.

Using the Monitor Controller

Using the Monitor Controller is very simple. The Monitor Controller UI only controls one box at a time, so if you have multiple boxes attached you need to select the box you want to control from the pop-up menu at the top of the window.


v.5 Monitor Controller UI

The monitor control window may be set to operate as a “Utility” Window (a “Utility” Window is a floating window that floats above all the other windows in the system — including windows from other applications). When configured in this way, the window will float above all other applications and will always be active (unless hidden). If you click the up-arrow button to the right of the box select popup menu so that it is highlighted, the window will operate as a Utility window. Click the button again (un-highlighting it) to return the window to operation as a normal floating window. The Key Command to hide and show the Monitor Control window is global and may be used to hide and show the Monitor Control window even when you are using other applications. When the Monitor Control window is functioning as a Utility window, it will remain visible and active even when MIO Console is hidden.

Once you have selected the box to control, you only have a few simple tasks to utilize the Monitor Controller.

Selecting an Input Source

The Monitor Source area will contain buttons representing all of your defined Monitor Sources. To select the input that will feed your Output Path, simply click on one of these buttons. The selected input will automatically be routed to the current monitor output. There are key-commands defined for the first eight monitor sources.

Selecting an Output Path

The Monitor Output area will contain buttons representing all of your defined Output Paths. To select the monitor output that will be fed by the selected Monitor Source, simply click on one of these buttons. The selected output will automatically be routed so that it receives signal from the currently selected input. There are key-commands defined for the first eight monitor output paths.

Adjusting the Monitor Level

To adjust the monitor volume, click and drag the Level Control Knob; drag the knob up to raise the volume, drag it down to lower the volume. You can also adjust the monitor level using two-finger trackpad scrolling, mouse-wheel, or mighty-mouse wheel when the cursor is placed over the Level Control Knob. If you know the exact value you wish for the monitor level, you can also click inside the Monitor Level Display and enter a numeric value directly.

Dimming the output

Sometimes, you may want to temporarily drop the output level significantly. For these situations, the Monitor controller includes a Dim button that will drop the output by 20dB when Dim is engaged. Click the Dim button to engage/disengage the Dim function.

Muting the output

Click the Mute button to mute/unmute the currently selected Monitor Output. The button appears grayed out when the output is muted.

Locking the output

Click the lock button to lock/unlock the monitor output level at its current setting. The button appears grayed out when the output level is locked. The Monitor Source and Monitor Output routing capabilities and mute functionality of the Monitor Controller function normally when the output level is locked.

Overriding the currently selected source

By clicking on the Source Override checkbox, you can switch the selected Monitor Source for the sources selected in the channel pop-up menus of the Source Override section. This can be useful if you find that you need to monitor something, or to A/B your normal Monitoring Sources, with a “one off” source. For example, you may have a number of Monitor Sources configured for analog audio streams, audio from your computer, and so on, but for one session only, you may need to monitor a signal streaming in digitally from the AES inputs. So in this situation, rather than configuring a Monitor Source, you might want to simply select the AES inputs in the Source Override channels.

Selecting Source Override Channels

Before you can use the Source Override checkbox, you need to use the channel pop-up menu to select a source for at least one channel.

You do not need to select sources for every pop-up menu in the Source Override section:

Note: You can not switch to a 7.1 surround source using Source Override.

The Monitor Controller as Floating Window

You can configure the Monitor Controller to be a floating window, also known as a “Utility Window” in official Mac OS X terminology. As a Mac OS X Utility Window, the Monitor Controller will remain “floating” on top of all other windows on screen, even if the MIO Console application is hidden. One advantage to setting the Monitor Controller to be a floating window is that you will always be able to quickly adjust the monitoring level regardless of how many other windows you have active. If you have an unexpected noise burst, this feature might just save your equipment—and your hearing!

Click the Up Arrow button in the upper right corner of the Monitor Control window to toggle between Utility and non-Utility mode.

Mini Monitor Controller Window

The full Monitor Control window requires a fair amount of screen space. For this reason, you can invoke a Mini Monitor Controller window, which offers nearly identical functionality in a window much smaller than the full Monitor Control window.


v.5 Mini Monitor Controller UI

As you can see, the Mini Monitor Controller offers identical buttons for Dim, Mute, and Lock, as well as the Utility Window button in the upper right corner. You can select among your configured Monitor Sources and Output Paths using the respective pop-up menus. You can invoke the Source Override section using the “S” button. You can adjust the volume level using the Level Control slider. Invoking the Configure window and choosing Source Override channels are the only functions that you cannot do from the Mini Controller window.

Switch between Monitor Controller and Mini Monitor Controller windows

You can toggle the display between the large Monitor Control Window and the Mini Monitor control window by clicking the Zoom button in the window’s title bar (the little green + pill).

Note: Here’s a way to conserve screen space while keeping the Monitor Controller available:

  1. Configure all your Monitor Sources, Output Paths, and Source Override channels in the Monitor Control window
  2. Toggle the window to the Mini Controller
  3. Click the Utility mode button
  4. Place the Mini Controller in an unobtrusive part of your monitor, such as one of the corners.
You now have a fully configured monitoring controller always available to you that requires a minimum of screen real estate.

Key Commands

The Monitor Controller defines key commands for each of its operations. You can change the default key commands with the Console’s Key Commands window. If you have a programmable HID device (like the Contour Shuttle Pro), you can assign appropriate key-commands to the buttons of the HID device to control the monitor controller. You can edit all the Key Commands in MIO Console.

Note: The key commands you have set for the monitor control commands are global; they work even if MIO Console is not the front-most application. This means that if you set up your HID controller to use the MIO Console key commands always (e.g. in global mode), you can control the monitor controller from the HID device or keyboard even when you are in another app (like iTunes, Logic, DP, the Finder, etc.).

The following table lists all the default key commands:

Default Monitor Control Key Commands

Command Key Sequence
Switch to/from Mini Controller ⌘⌥⌃F (Command + Option + Control + F)
Volume Up ⌘⌥⌃↑ (Command + Option + Control + up arrow)
Volume Down ⌘⌥⌃↓ (Command + Option + Control + down arrow)
Toggle Dim ⌘⌥⌃D (Command + Option + Control + D)
Toggle Mute ⌘⌥⌃M (Command + Option + Control + M)
Toggle Window Visibility ⌘⌥⌃V (Command + Option + Control + V)
Select Monitor Source 1 ⌘⌥⌃1 (Command + Option + Control + 1)
Select Monitor Source 2 ⌘⌥⌃2 (Command + Option + Control + 2)
Select Monitor Source 3 ⌘⌥⌃3 (Command + Option + Control + 3)
Select Monitor Source 4 ⌘⌥⌃4 (Command + Option + Control + 4)
Select Monitor Source 5 ⌘⌥⌃5 (Command + Option + Control + 5)
Select Monitor Source 6 ⌘⌥⌃6 (Command + Option + Control + 6)
Select Monitor Source 7 ⌘⌥⌃7 (Command + Option + Control + 7)
Select Monitor Source 8 ⌘⌥⌃8 (Command + Option + Control + 8)
Select Monitor Output 1 ⌘⌥1 (Command + Option + 1)
Select Monitor Output 2 ⌘⌥2 (Command + Option + 2)
Select Monitor Output 3 ⌘⌥3 (Command + Option + 3)
Select Monitor Output 4 ⌘⌥4 (Command + Option + 4)
Select Monitor Output 5 ⌘⌥5 (Command + Option + 5)
Select Monitor Output 6 ⌘⌥6 (Command + Option + 6)
Select Monitor Output 7 ⌘⌥7 (Command + Option + 7)
Select Monitor Output 8 ⌘⌥8 (Command + Option + 8)

Configuring the Monitor Controller

Click the “Configure…” button to open the Configuration Dialog for the Monitor Controller. See the next sections for details.

Configuring Monitor Sources

To access the Configuration Dialog, click the “Configure…” button in the Monitor Controller window:


The Monitor Controller’s Configuration Dialog sheet will open:


To Add a New Monitor Source:

  1. Click the ‘+’ button in the Monitor Sources pane.
  2. The Add Monitor source dialog appears:


  3. Enter the name of the new Monitor Source.
  4. Select the Bus Type of the new Monitor Source.
  5. Click OK. The new source will appear in the Monitor Source List:


  6. Click the pop-up menu in the “Signal Source” column for the Left Channel of the source. Select the appropriate physical source channel from the list. This includes Physical Input and DAW channels.
  7. Repeat step 5 for each of the channels that make up the bus.
  8. If you decide (now or later) that you want to change the name of the Monitor Source, you can edit the name in the list (you can double-click the name to edit it).
  9. Repeat from step #1 for each source you want to add.

To Remove a Monitor Source:

  1. Click the source(s) you want to remove in the Monitor Source List.
  2. Click the ‘-‘ button.

To change the order of the Monitor Sources:

  1. Click the source(s) you want to move in the Monitor Source List.
  2. Click the up-arrow button to move the selected sources higher in the list.
  3. Click the down-arrow button to move the selected sources lower in the list.

Configuring Monitor Output Paths

Monitor Output paths are more configurable than Monitor sources; besides configuring the actual output channels in the path, you can also define the level standard used by the output channel (if your hardware supports that) and per-channel level offsets. This is useful for compensating for gain structure differences in output paths on a per-channel basis. For example, if you have some balanced amps and unbalanced amps in your system you can define different level standards for them, or if different amps have different sensitivities, you can define different per-channel offsets. The per-channel level control and per-channel output type controls are accessed from the Monitor Controller configuration dialog.

The Monitor Controller also supports per-path level calibration to allow you to calibrate and normalize levels between different output paths. This can be used to set your nominal 0 dB level to match a monitoring standard or to normalize levels between multiple monitor paths.

To access the Configuration Dialog, click the “Configure…” button in the Monitor Controller window:


The Monitor Controller’s Configuration Dialog sheet will open:


To Add a New Monitor Output Path:

  1. Click the ‘+’ button in the Monitor Paths pane.
  2. The Add Monitor Output dialog appears:


  3. Enter the name of the new Monitor Output Path.
  4. Select the Bus Type of the new Monitor Source.
  5. Don't worry about the “Type” pop-up menu — it is reserved for future use. All you need to know is that the Type must be “Control Room” for the Monitor Controller to work correctly.
  6. Click OK. The new source will appear in the Monitor Paths List:

  7. Click the pop-up menu in the “Output” column for the Left Channel of the Monitor Path. Select the appropriate physical destination channel from the list.
  8. Repeat step 7 for each of the channels that make up the bus.
  9. If you decide (now or later) that you want to change the name of the Monitor Output Path, you can edit the name in the list. Double-click the name to edit it.
  10. Repeat from step #1 for each Monitor Output Path you want to add.

To Remove a Monitor Output Path:

  1. Click the output path(s) you want to remove in the Monitor Paths List.
  2. Click the ‘-‘ button.

To change the order of the Monitor Output Path:

  1. Click the output path(s) you want to move in the Monitor Paths List.
  2. Click the up-arrows button to move the selected output paths higher in the list.
  3. Click the down-arrows button to move the output paths lower in the list.

To change the per path Calibration:

The Monitor Output Path calibration control is accessed via a contextual popup menu on the main Monitor Controller level control knob:

v.5 Monitor Control Calibration

To Calibrate a Monitor path:
  1. Select the output path to calibrate (Mains in the screenshot above).
  2. Adjust the level knob to generate the desired reference level at the output.
  3. <control>-click (or right-click a multi-button mouse or trackball) the knob to popup a contextual menu.
  4. Select the Set Path Calibration item.
  5. The calibration will be set for the path, and the level control will be adjusted to read 0.0.
To Remove Calibration from a Monitor path:
  1. Select the output path to calibrate (Mains in the screenshot above).
  2. <control>-click the knob to popup a contextual menu.
  3. Select the Remove Path Calibration item.
  4. The calibration will be removed from the path, and the level control will be adjusted so that the output level does not change.

Monitor Controller FAQ:

  1. Why are some of my output trim controls greyed out in the “Analog I/O Control” pane of the MIO Console window?

    A: When you have assigned an output as part of a Monitor Path, the monitor controller takes control of that channel for the purpose of controlling its output level. Since the monitor controller is controlling the channel level, it would not make sense for the trim knob in the Analog I/O Control pane to also change that setting. As a result, the corresponding control in the “Analog I/O Control” pane is disabled and its setting automatically updated as you adjust the Monitor Controller. If you remove that channel from all Output Paths in the monitor controller, that output will be restored to manual control in the “Analog I/O Control” pane, and its trim knob will no longer be greyed out.

  2. Why are some of my output routing controls greyed out in the Output Patchbay of the “Mix/Output Routing” pane of the MIO Console window?

    A: When you have assigned an output as part of a Monitor Path, the monitor controller takes control of that channel for the purpose of controlling its routing. Since the monitor controller is controlling the channel, it would not make sense for you to change the routing independently. As a result the corresponding routing control in the “Mix/Output Routing” pane is disabled and “greyed out”. Its setting is automatically updated by the monitor controller as you make changes, and if you remove that channel from all monitor paths, it will be restored to manual control in the “Mix/Output Routing” pane.

  3. Why is the Monitor Level Control and Dim button greyed out and disabled?

    A: There are two reasons that the Level Control can be disabled. The first reason is that you have locked the Level Control with the “Lock” button. The second reason is that there is no currently selected Output Path.

  4. Why does selecting one Monitor Path mute all the other Monitor Paths?

    A: Control Room Monitor Paths are exclusive; only one is active at a time. As a result, selecting one Monitor Path will automatically mute all the other paths. If you want an output to be active all the time, route it manually in the patchbay — don’t add it to the Monitor Controller. This would be appropriate for an aux send, for example.

  5. I can’t figure out how to configure the Monitor Sources or Output Paths from the mini Monitor Control Window. How do I do it?

    A: You can’t configure the Monitor Controller directly from the mini Monitor Control Window. You need to switch back to the large Monitor Control Window first by clicking the Zoom button in the mini Monitor Control Window.

  6. I can’t figure out how to configure the Source Override sources from the mini Monitor Control Window. How do I do it?

    A: You can’t configure the Source Override directly from the mini Monitor Control Window. You need to switch back to the large Monitor Control Window first by clicking the Zoom button in the mini Monitor Control Window.

 Routing Applications for 2d Expanded Units

Routing Applications for 2d Expanded Units

Routing Applications

This document provides a hands-on description of how to configure the mixer and router for a somewhat complex tracking situation. This document is only for 2d Expanded units. You can compare this document to corresponding document (Routing Applications for Unexpanded Units) for un-expanded units to see how much easier the process has gotten with the 2d Card and the v.5 mixer.

EXAMPLE SETUP

2882.png
Example ULN-2 Setup

In this section we will go step by step through an application that is possible with a ULN-2 configured as shown in illustration Example ULN-2 Setup above. After reading this section you should have a feel for how to apply Mobile I/O’s routing and mixing capabities.

EFFECTS FOR TRACKING

Many performers (especially singers) find it easier to get a good take if they have some sweetening effects in their headphone mix while tracking. These will usually be temporary effects and will not necesarily make it into the final mix but they can make the difference between a good take and a bad take.

In previous versions of the Mobile I/O, we would have to use an external effects device connected to Mobile I/O’s AES ports to add some reverb to the headphone mix we make in MIO Console, but with the introduction of the 2d Card, we can add reverb “In the Box”. You can still use the AES ports to add other effects if desired.

SETTING UP THE ROUTING

The first thing we need to do is configure our mixers and routing. For this setup we will use two mix busses in MIO Console: one for the actual headphone mix and one for the send mix to the reverb. To do this we need to bring up the Mixer Config sheet for the v.5 mixer:

2882.png
Mix Config sheet

Add two busses to the mixer by clicking the + button in the Mix Busses section twice. Double click the first bus to name it, and tab to name the second bus. Also, set the Reverb bus’s Bus Mode to Aux Bus:

2882.png
Newly Created Busses

After you have created the busses, you still need to assign the desired inputs to the Monitor bus. To do this, first select the Monitor bus in the Mix Busses list and then just click the check boxes for the inputs you need in the Selected Mix Bus Configuration list

In addition, if you would like to use any of the other inputs (AES or ADAT), you can simply add them to the bus by clicking the proper check boxes in the list.

You don't need to add any of the inputs to the Reverb bus — we’ll do that with sends in the mixer.

Click the Configure button, and MIO Console will configure the basic mixer for you:

2882.png
Basic Mixer Configuration

Now we need to add sends to get the audio from the inputs to the Reverb bus. This can be done in bulk by selecting the desired input strips and then inserting the send in one of the insert slots.

First select the strips:

2882.png
Selected Strips

Then insert the send:

2882.png
Inserting Send to Reverb Bus

After you have done this, the Sends window will appear, and each strip will have a send inserted:

2882.png
After Inserting the Sends

As you can see, you now have a fader and panner for each send that you can use to set the level and pan of the signal into the reverb. The lavender colored P buttons on each send strip make the send strip Post Fader when enabled, and Pre Fader when disabled.

We still need to route the Monitor Mix to the headphones. This is done directly from the Master Fader strip output popup at the very bottom of the Monitor strip:

2882.png
Making the Output Assignment

Finally, we need to add a HaloVerb reverb plugin to the Reverb bus return:

2882.png
Selecting the Reverb Plugin
Once you insert the plugin, you will see the HaloVerb UI with the Factory Default settings:
2882.png
Factory Default Settings
Now, for the configuration we have set up, with a dedicated return for the reverb that will be mixed back in with the Monitor bus, we really don’t want to have the dry signal of the Reverb bus mixed back in; rather we want to have the 100% wet signal, so we need to turn down the Direct dB parameter to mute the dry signal:
2882.png
100% Wet Default Settings

Now everything is basically configured. You can set up the dry headphone mix on the main faders in the Mixer window. Once we have basic balances set, you can switch to the Send window. In the Send mixer we can unmute each channel and set its send level to the reverb. If you want it to match the levels of the dry headphone mix, you can ensure that all the sends are Post Fader and all the send faders are set to unity gain. If you would rather set the reverb send levels independently of the dry mix, you can make the sends Pre Fader.

Tweak the HaloVerb settings to taste. Notice that even if you set the Direct dB parameter back to 0 dB there is no phasing introduced, because the v.5 mixer is absolutely phase coherent, even with routing between multiple busses.

Your final configuration will look something like this:

2882.png
Final Configuration

If you want to use this example as a tracking set up, choose Save As... from the file menu. This brings up a standard Macintosh save dialog and allows you to save the entire state of MIO Console as a setup document.

Controlling Multiple Monitors

You can use the Monitor Controller features of Mobile I/O to set up multiple monitor paths and easily switch between them. The Monitor Controller is an intelligent router and volume controller that was designed specifically to manage monitoring systems.

For this example we will work with a 2882 and have two sets of powered monitors connected to the Mobile I/O. The first set will be connected to Analog Outputs 1-2 and the second to Analog Outputs 3-4.

2882.png
Example Setup

We will start with the “2882 Basic with Monitor Control” template as that configuration gets us most of the way there. The configuration looks like:

2882.png
2882 Basic with Monitor Control
The starting template is appropriate for recording with a 2d Expanded 2882 and playing stereo material through the box. It is set up to allow you to do zero-latency monitoring without additional configuration. It also contains a configured monitor controller for monitor switching and volume control.

This template configures a single 2882 with:

However, the template does not provide support for our scenario of two sets of stereo monitors. So we can quickly add that to the configuration. The Monitor Controller configuration in the template looks like:

2882.png
Template MC Configuration

The configuration includes a DAW Direct source that routes directly from the DAW 01 & DAW 02 channels from the computer, and the Main Stereo bus is also available as a source for the Monitor Controller.

The configuration also includes two Monitor Paths for output. One, named “Main Monitors”, routes out to Analog 1+2, and the other, named “Cans”, routes out to the Cans connector on the front panel.

This is pretty close to what we want, but we need to have another Monitor Path for the “Small” monitors, and we want to rename “Main Monitors” to “Big”.

So, first we add another Monitor Path by clicking the + button in the Monitor Paths section:

2882.png
Add Small Monitor Path

Then select the output channels for the path (click the popup for the Left Channel); hold down the option key to select successive channels in one step:

2882.png
Select Path

After you make the selection the path will be configured:

2882.png
Channels Selected

You can close the disclosure arrows for the Monitor Paths list:

2882.png
Closed List

And move the Small Monitor Path to be the second item by clicking the up arrows button:

2882.png
Reordered List

And rename the “Main Monitors” Path to “Big” by double clicking the path, and typing the new name. When you are done it will look like:

2882.png
Final MC Config

After you click the OK button, the Monitor Controller window will look like this:

2882.png
Final Monitor Controller

We want to make sure that both Analog 1+2 or Analog 3+4 has signal, but not both at the same time; sound should only come from one set of speakers at a time. That’s precisely what the Monitor Controller will do for us; sound will only be routed to one of the Monitor Paths at a time, and the unselected Monitor Paths will be muted.

To switch between the Big monitors and the Small monitors, all you have to do is click on the path you want to be active in the output selector bar right above the output gain control knob.

If you want to use this example as a tracking set up, choose Save As... from the file menu. This brings up a standard Macintosh save dialog and allows you to save the entire state of MIO Console as a setup document.

Conclusion

The preceding examples should give you a sense of the possibilities that are enabled by the routing and mixing features of Mobile I/O. While this is just a starting point, we have covered all of the basic operations required to manipulate Mobile I/O with complex routing. You should be able to build upon these scenarios to construct routings that suit your needs and your workflow.

 Routing Applications for Unexpanded Units

Routing Applications for Unexpanded Units

Routing Applications

This document provides a hands-on description of how to configure the mixer and router for a somewhat complex tracking situation. This document is only for un-expanded units (Mobile I/O units that do not have a 2d Card installed). If your unit has a 2d Card installed, the configuration process is very different (and much simpler).

For information about Routing Applications with a 2d Expanded unit, please refer to the Routing Applications for 2d Expanded Units document.

Clocking Considerations

There are five ways that you can clock Mobile I/O:
  1. Internal
  2. Digital (AES/SPDIF)
  3. Optical (ADAT/TOSLINK)
  4. Wordclock
  5. 256X

Each choice is appropriate for a particular situation. In general, you should use Internal clock if you can because you are likely to find that it makes your other digital gear sound better. However, there are some devices that either must be the clock master or work better if they are the clock master. For these devices, choose the clock port which is most appropriate for the device.

For example, many DAT machines distribute clock over their digital audio connection (AES or S/PDIF). For these machines you would connect to the AES or S/PDIF port on Mobile I/O and choose DigIn (44.1/48) for single rate or DigIn (88.2/96) for double rate in the MIO Console clock source popup. For more information about configuring the clock source, see “System Controls”.

SINGLE SPEED CLOCKING VS. DOUBLE SPEED CLOCKING

The difference between single speed (1x) and double speed (2x) clocking is handled as a fundamental mode change in Mobile I/O. The sample rate does not vary continuously between 48k and 88.2k but changes discontinuously when the double speed mode is set in the hardware.

When Mobile I/O is running on internal clock, this mode change is handled transparently when you specifiy what sample rate the box should use.

When Mobile I/O is running on external clock, it cannot determine in advance which mode to use, so you need to tell it what mode is appropriate for your external clock source. You communicate this information by selecting the proper external clock source. Each external clock source comes in two different flavors:

where the “xxxx” corresponds to the actual clock source. If you will be clocking off of a 1x source (e.g. 32kHz–50kHz sample rate) then choose the xxxx(44/48) clock source. On the other hand, if you will be clocking off a 2x source (e.g. 64kHz–100kHz sample rate) then you will choose the xxxx(88/96) clock source. If you choose the wrong flavor, Mobile I/O will not lock properly. When you select the flavor, it tells Mobile I/O which mode it has to run in, and the Mobile I/O can lock to the external clock in the appropriate frequency range.

EXAMPLE SETUP

2882.png
Example ULN-2 Setup

In this section we will go step by step through an application that is possible with a ULN-2 configured as shown in Example ULN-2 Setup above. After reading this section you should have a feel for how to apply Mobile I/O’s routing and mixing capabities.

WHICH DEVICE IS THE CLOCK MASTER

The first consideration involved with a setup like this is which device should be the clock master. If the external device is happy slaving to Mobile I/O then make Mobile I/O the clock master by setting the clock source to internal in MIO Console. Set the external device to lock to its digital input, or wordclock if the external device has a wordclock input. If you use wordclock, make sure to connect a 75 Ohm BNC cable between Mobile I/O and the external device.

If you find that you are getting clicks and pops in the audio coming from the external device, try clocking in the other direction. Set the external device to use its internal clock and set the Mobile I/O clock source to digital or wordclock. Remember to choose the appropriate clock rate (single or double) for the sample rate you will be working at.

Now that all of the digital devices are playing nicely with each other, let’s take a look at some possible applications with this setup.

EFFECTS FOR TRACKING

Many performers (especially singers) find it easier to get a good take if they have some sweetening effects in their headphone mix while tracking. These will usually be temporary effects and will not necesarily make it into the final mix but they can make the difference between a good take and a bad take. In this example, we will be using an effects device connected to Mobile I/O’s AES ports to add some reverb to the headphone mix we make in MIO Console.

SETTING UP THE ROUTING

The first thing we need to do is configure our mixers and routing. For this setup we will use two mixers in MIO Console: one for the actual headphone mix and one for the send mix to the reverb. To do this we need to bring up the Mix/Output Routing view in MIO console

2882.png
Configuring the Matrix
In its default state, the WIDE mixer is configured with only the Analog Inputs assigned to the mix busses, so we need to add the Digital Inputs. To do this we simply click in the assign tiles under the Digital Inputs for the mix busses we want. For this example we will use mix busses 1 and 2 for the Monitor mix and mix busses 3 and 4 for the effects send mix.
2882.png
Naming the mix busses

We’ll also assign DAW 1 and 2 to mix busses 1 and 2 for DAW playback. We can name the mix busses by clicking on the mix bus label. This will show the matrix configuration window (Naming the mix busses).

We’ll name the headphone mix Monitor L/R and the send mix Effects Send L/R. You must name each mono bus separately, and the Console will combine the names automatically. You can use the tab key to increment through the mix busses as you name them. When you’re finished, hit the return key to close the configuration window.

The mixer bus names have propagated through MIO Console which means they will now show up in the Output Patchbay and the Mixer view.

Note that Digital L/R is assigned to the Headphone mix but not the Effects send mix. This is critical because Digital L/R is the effects return, and we definitely do not want to assign the return to the send. If we do, we’ll get feedback and no one will be happy, especially the singer.

To route the mixers to their intended destinations we use the Output Patchbay. We want the Headphone mix to feed the front panel headphone output of Mobile I/O so we’ll set the Monitor section of the Output Patchbay to take its input from Monitor L/R

2882.png
Making the Output Assignment

The Effects Send mix should feed the AES outputs so we’ll set the Digital section of the Output Patchbay to take its input from Effects Send L/R.

The routing is now set. We can save the setup for the next tracking session using the Matrix Parameters and Patchbay Parameters popup menus (see “Parameter Popup Controls” on page 50 for details).

Now we’ll switch to the Mixer View.

2882.png
Headphone Mixer

Notice the Mix Tabs on the left are now labeled Monitor L/R and Effect send L/R. Also notice the fader marked Daw 1/2 on the right. We’ll use this fader to control the playback level of the DAW tracks we want to overdub against.

If you have signals running to Mobile I/O you should see activity on both the input meters and the mixer meters.

By default the mixer’s faders are are at unity gain and the mutes are engaged. We can unmute all the mixer channels by holding the option key and clicking one of the mutes. All of the channels will unmute. But for the time being we’ll keep the Digital L/R fader muted.

Now we can set up a dry headphone mix. Once we have basic balances set, we can switch to the Effects Send mixer by clicking its mix tab. In the Effects Send mixer we can unmute each channel and set its send level to the reverb, or we can use the Mixer parameters popup (see “Mixer Pane Tabs” on page 36) to copy the mix from the Headphone mixer to use as a starting point.

To copy mix parameters from one mixer to another:

  1. Click and hold the mix tab for the mixer you want to copy from. A popup menu will appear:
    2882.png
    Mixer parameters popup
  2. Choose Copy Parameters
  3. Now click and hold on the mix tab you want to copy to
  4. Choose paste parameters
  5. The mix parameters have now been copied

Now we can can go back to the Monitor mixer and unmute the Digital fader. The reverb should now be heard in the headphone mix.

If you want to use this example as a tracking set up, choose Save As... from the file menu. This brings up a standard Macintosh save dialog and allows you to save the entire state of MIO Console as a setup document.

The preceding examples should give you a sense of the possibilities that are enabled by the routing and mixing features of Mobile I/O. While this is just a starting point, we have covered all of the basic operations required to manipulate Mobile I/O with complex routing. You should be able to build upon these scenarios to construct routings that suit your needs and your workflow.

CONTROLLING MULTIPLE MONITORS WITH THE 2882

We can use the routing snapshot features of Mobile I/O to set up multiple monitor paths and easily switch between them.

EXAMPLE SETUP

2882.png
2882 Example Setup

For this example we will have two sets of powered monitors connected to Mobile I/O. The first set will be connected to Analog Outputs 1-2 and the second to Analog Outputs 3-4.

In the routing window we’ll set up a monitor mixer on busses 5 and 6 and name it Monitor L/R. We’ll assign Analog 1-8, ADAT 1-8, Digital L/R and DAW 1/2 to the monitor mixer. This will allow us to monitor playback from a CoreAudio application as well as our live inputs.

2882.png
Setting up the monitor mixer

Now in the Output Patchbay we’ll assign Monitor L and Monitor R to Analog 1 and Analog 2. This creates the routing for our Big Monitors. We’ll also make sure to assign an output pair that we know won’t ever have signal on it to Analog 3 and Analog 4. We’ll use Daw 17 and Daw 18.

2882.png
Setting the Monitor path

We want to make sure that Analog 3 and Analog 4 won’t ever have signal in this setup because the Small Monitors are connected to Analog 3 and Analog 4 and we don’t want to have random channels playing through the Small monitors while we are listening to the Big Monitors.

Next we’ll save this routing using the Patchbay parameters popup menu. But first we’ll create a new category by choosing Create new category... from the Patchbay Parameters popup.

A dialog will appear:

2882.png
Naming a category

We’ll name the category “Monitors” and click OK.

Now we can save the setup into the Monitors category by choosing Save As... from the Patchbay Parameters popup. A dialog will appear:

2882.png
Saving a setting

We’ll choose the Monitors category from the category popup, name the setup “Big Monitors” and click OK. The setup has now been saved.

Now we can set up the routing for the Small monitors. To do this we simply assign the monitor mixer to Analog 3 and Analog 4 and assign DAW 17 and DAW 18 to Analog 1 and Analog 2.

2882.png
Small Mixer setup

We’ll save the setup into the Monitors category as “Small Monitors” using the Parameter Patchbay popup. Now we have a monitor switching system. To switch monitors, simply choose the monitor path you want using the patchbay popup window.

2882.png
Choosing the Monitor Path
 Troubleshooting Guide

Troubleshooting Guide

COMPUTER DOES NOT SEE MOBILE I/O

If you attach Mobile I/O to your computer, and the computer is unable to communicate with the Mobile I/O hardware there are five basic possibilities for the source of the problem:

  1. The Mobile I/O is not powered up
  2. The Software is not installed properly
  3. The FireWire bus did not reset correctly
  4. The FireWire cable is bad
  5. The FireWire hardware has been damaged

MOBILE I/O IS NOT POWERED UP

The first thing to check is that the Mobile I/O is, in fact, powered up.

If Mobile I/O is powered up and booted properly, the Power, Sample Rate, and Locked front panel indicators will be illuminated. If these indicators are not illuminated, the Mobile I/O is not powered properly or the unit’s firmware has been corrupted. If you determine that you are powering Mobile I/O properly and the indicators are not illuminated, you will need to contact Metric Halo support.

If you are bus powering the Mobile I/O, there is a possibility that you have overloaded the power rating of the power source. Please see the troubleshooting section “Not enough power on the Bus” for details on troubleshooting this problem.

If the Mobile I/O is properly powered, then check the next possibility.

SOFTWARE IS NOT INSTALLED PROPERLY

In order for the computer to properly communicate with the Mobile I/O, the various components of the driver software need to be installed correctly. If the software is not installed correctly, the communication between the computer and Mobile I/O will fail in various ways. If the MobileIODriver.kext is not properly installed in the /System/Library/Extensions folder of your computer, you will not be able to use the Mobile I/O for audio and you will not be able to control the sample rate or clock source of the Mobile I/O with the computer.

THE FIREWIRE BUS DID NOT RESET CORRECTLY

When a device is plugged into the FireWire bus, a FireWire bus reset occurs automatically. The bus reset interrupts bus activity and reconfigures the bus so that all devices on the bus become aware of all the other devices on the bus. Sometimes the reset does not complete successfully, and the bus becomes partially hung. In this case, the “FireWire” indicator on the front panel of the Mobile I/O will not be illuminated. When the “FireWire” indicator on the front panel is not illuminated, the Mobile I/O cannot transport audio over the FireWire bus.

Generally, this condition can be fixed by disconnecting the Mobile I/O from the bus and reconnecting it.

If the disconnect/reconnect cycle does not fix the problem, another device on the bus may be interfering with the proper operation of the bus. If you have other devices on the bus, try disconnecting them from the bus and only using the Mobile I/O.

If removing other devices from the bus solves the problem, it is likely that there is a problem with either one of the devices you removed or with one of the cables connecting the devices. You’ll need to isolate the problem component.

If removing the other devices from the bus does not fix the problem, check the next possibility.

THE FIREWIRE CABLE IS BAD

Metric Halo provides two high-quality overspec’ed FireWire cables for use with Mobile I/O and we recommend you use them. For various reasons you may decide to use other cables than the ones provided by Metric Halo. Under ideal circumstances all FireWire cables will provide years of service. However, cables will and do go bad. Cable failures can be difficult to track down. If you are experiencing problems with connecting or bus powering Mobile I/O you should try swapping the cable with another known-good cable.

If the FireWire cable is not the source of the problem, check the next possibility.

THE FIREWIRE HARDWARE HAS BEEN DAMAGED

If all else fails, it may be that the FireWire hardware on either the Mobile I/O or the computer has been damaged. While this is an exceptionally rare occurrence, it is a possibility. The FireWire hardware can be damaged in the following ways:

  1. If you insert a FireWire cable into a port upside down, it will damage the FireWire port and/or the connector. It is difficult to insert the connector upside down, but it is possible to force it. Never force a FireWire connector!
  2. It requires significant pressure, but it is possible to force a FireWire connector over a male XLR connector pin. If you do this, the connector will be shorted and it will destroy the port on the other end. Again, never force a FireWire connector.
  3. Some devices that are bus-powerable and conform to the IEEE1394 standard will return power to the remote FireWire port if a power ground fault occurs. If the remote port is protected against this situation, nothing will happen. If the device does not use bus power, nothing will happen. But, if the device is fully compliant, uses bus power, and the remote device is not protected and supplies a high enough voltage on the bus, the remote device port will self-destruct.

If the FireWire hardware on the computer has been damaged, it will not communicate with any FireWire devices. Be sure that you are not checking this case with a bad cable, as a bad cable can make it seem like the FireWire hardware has failed since it will consistently keep devices from connecting properly to the computer. If the computer is damaged, you will need to contact the manufacturer for a repair or, as a stopgap measure, you can use a third-party FireWire adapter card. Metric Halo also recommends the use of a third party FireWire hub between the computer and any FireWire devices. Any potential issues will damage the hub, not the computer, saving time and money in the long run.

If the FireWire hardware on the Mobile I/O has been damaged the MIO will not communicate with any other devices. In this case, please contact Metric Halo support for help in getting your Mobile I/O hardware repaired.

GROUND LOOPS

Audio systems, in general, are susceptible to ground loop problems. Digital Audio Interfaces for computers are even more susceptible to grounding issues since they must interface with the computer’s system ground, which tends to be much more dirty than the ground used by audio gear. By taking care when you connect the various components of your audio system you can avoid the hums, buzzes, and noises that characterize ground loops and other grounding problems.

First of all, most grounding issues go away if you utilize balanced interconnects between your audio gear. Balanced interconnects inherently reject ground differentials and common mode interference introduced by grounding problems. Balanced connections are not much more expensive than unbalanced connections and solve so many problems that if both ends of the connection support balanced interconnect, you should not even consider using unbalanced cables.

You may get the idea that we hate unbalanced connections. You’re right. We do. You should too.

If you have to use unbalanced connections, or if any ground-related problems remain, you will find that the key to the issue is ensuring that you have a common hard ground between all the gear that you are interfacing. This is commonly referred to as a technical ground. A technical ground is characterized by a consistent low impedance path between each device and a common reference ground, ideally connected directly to earth ground. The above is sometimes difficult due to electrical wiring problems in the house, studio, or stage you are using. In the extreme case, you may need to hire a qualified electrician to untangle and correct electrical service problems in your working environment.

Unbalanced connections are a fact of life when interfacing with guitar amps, and, paradoxically, guitar amps are extremely sensitive to grounding issues since they use so much gain to achieve the effect of a “Guitar Amp”. If you will be interfacing with guitars and guitar amps, you need to be very careful about grounding.

Common electrical wiring approaches to residential installations, and sub-par studio and stage installations use daisy-chained grounds for ease of installation and economy. Unfortunately, daisy-chained grounds can introduce signficant ground differentials between sockets, and these differentials can vary depending on other loads (like refrigerators, TV’s and other household appliances) on the circuit.

Other problems with electrical service installations are improper wiring of power phases to the three-phase service and improper connections between the safety ground and hot legs of the three-phase service. These types of problems tend to be characterized by loud 60Hz hums in the audio system. Unfortunately, these types of problems extend well beyond noise in your audio system to genuine safety hazards. If you determine that your electrical wiring has problems beyond a simple daisy-chained ground, you should consult a licensed electrician immediately, as ignoring these problems can damage either you or your gear.

If you do not have a well implemented technical ground, you will want to ensure that all of the devices in your audio system are plugged into the same phase and same ground. You can generally accomplish this by running all your gear off of the same socket (using a power strip or power conditioner) if your gear uses less power than is supplied by a single circuit from your premises’ wiring (generally 10-15 amps in residential installations and 20 amps in commercial installations).

It is usually a bad idea to put some devices in your system on a power conditioner and other devices on a separate strip, socket or conditioner, unless you have a technical ground. The power conditioner can introduce a ground differential.

The power supply provided with Mac laptops does not have a hard ground. This means that if the laptop is plugged in, it will dump high frequency buzz into the ground. That ground is shared with the Mobile I/O Firewire cable. If Mobile I/O will be connected unbalanced to other audio gear, the ground buzz can contaminate the signal if the Mobile I/O is not hard-grounded to the same ground as your other audio gear. To hard ground the Mobile I/O you will need to use a 3-pin power cable on the Mobile I/O power supply and power the Mobile I/O with the power supply. Plug the 3-pin IEC power cable into the same circuit and same ground as your other gear.

On the other hand, if you are encountering ground loop problems while operating with the Mobile I/O’s power supply, you may find that lifting the Mobile I/O’s ground resolves the problem. This can be accomplished by using a 2-Pin IEC cable (without the third ground pin), or by using a ground lift block (generally available in hardware stores, also known as a 3 pin to 2 pin converter). In general, it is better to resolve the fundamental grounding problems in your system, but this is a quick fix that may help. There are no hard and fast rules for solving this type of problem other than fixing the fundamental grounding issues, so if you go this route, you will have to experiment with lifting various grounds in your system until you find the magic combination. Or switch to balanced interconnects.

Finally, the Apple Cinema Display has a known issue with its backlight dimmer. If you run the Apple Cinema Display with its backlight at anything other than full brightness, the backlight dimmer will introduce a midrange buzz into the system ground which will appear in unbalanced interconnects (input and output) with Mobile I/O. This issue affects other devices that connect to the computer’s system ground. The work around is to run the display at full brightness, or use balanced interconnects.

FIRMWARE UPDATE PROBLEMS

For details on updating the firmware of Mobile I/O refer to Appendix 1.

It is possible for firmware updates to “not take”. This appears to be related to DSP loading issues in the Mobile I/O, other devices on the FireWire bus, and the state of the FireWire system software on the Mac. If you have problems with updating the firmware, try the following procedure:

  1. Remove all devices from the FireWire bus
  2. If your Mobile I/O is using external power, disconnect the power
  3. Reboot your computer
  4. Attach the external power supply to the Mobile I/O while holding down the front panel Mute button; this will boot the Mobile I/O into the safety boot firmware
  5. Connect the Mobile I/O to your computer
  6. Run the firmware updater

Since Mobile I/O implements safe-boot and safe firmware update, you should always be able to use this procedure to update the firmware, even if something goes horribly wrong (like losing power during an update).

BUS POWERING MOBILE I/O

If you are bus powering the Mobile I/O, there is a possibility that you have overloaded the power rating of the power source.

NOT ENOUGH POWER ON THE BUS

While all Macintosh computers with built-in FireWire supply bus power, some models do not provide enough power on the bus to power Mobile I/O. If this is the case, you will generally find that the Mobile I/O will boot on initial connection, but will then lose power or will reboot repeatedly after a short period of operation.

Newer Intel Macs have implemented an in-rush current limit circuit breaker that may trigger when powering a Mobile I/O unit. This will prevent the computer from booting the unit, but not from powering it. So you can use an external supply or battery to get the unit powered up and then use the computer’s bus power to provide power to the Mobile I/O.

Some Mac models provide enough power if they are plugged into the wall, but will not provide enough power while running on batteries. If the computer does not provide enough power, you will need to use an external power source with Mobile I/O.

The external power supply provided with Mobile I/O is the perfect solution if you are using Mobile I/O in an environment where ac power is available. The external power supply will actually provide power to the bus and can be user to power other bus-powered peripherals (see other bus powered devices).

If AC power is not available, you will need to use an external battery-based power source to power Mobile I/O. Any source that provides 9V-30V and can support 12-15W of power consumption will work well with Mobile I/O. Check with Metric Halo for specific recommendations.

When using an external battery source, DO power Mobile I/O directly from the battery – not through an inverter. DON’T power the computer with an external battery and use the computer to power Mobile I/O; doing so will not resolve your bus power problems, and it will give you more limited run times. If you need to use the external battery with the computer use two batteries or split the DC supply at the battery and power both the Mobile I/O and the computer.

OTHER BUS POWERED DEVICES

Mobile I/O consumes enough power that it is very unlikely that you will be able to successfully bus power Mobile I/O and any other bus-powered device (except for a hub) from the same computer. If you plan on using other bus-powerable devices with your computer, you will need to either self- power your other devices or self-power the Mobile I/O. It is probably best to use the Mobile I/O’s power supply in this situation since Mobile I/O will then provide approximately 30 Watts of power to the bus (roughly 3x what most Portable Mac’s will supply). This will allow you to power all the rest of your devices without any concern of running out of power.

 Using the ULN-2 Hardware

Using the ULN-2 Hardware

ULN-2 Front Panel

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ULN-2 Front Panel

The ULN-2 front panel provides ten-segment metering for the 2 analog inputs and the main outputs as well as knobs and switches to control the input, monitor, and headphones sections. The meters are fast PPM peak reading meters with auto-resetting peak holds.

Each input channel has the following controls:

The front panel also provides ULN-2 system status at a glance:

The ULN-2 front panel provides access to the level control knobs for headphones and for the monitor outs. The headphone output jack is on the front panel and the monitor output jacks are located on the back of the unit.

The headphone output jack is a TRS 1/4” jack that provides the Left Channel on the tip, the Right Channel on the ring and the ground return for the two channels on the sleeve. The Monitor output jacks are balanced TRS connectors.

ULN-2 Rear Panel

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ULN-2 Rear Panel

The ULN-2 rear panel features:

Making connections to the ULN-2

There are six classes of connections you can make to the ULN-2 hardware:

  1. Analog Audio
  2. Copper-based Digital Audio
  3. Optical-based Digital Audio
  4. Clock Sync
  5. FireWire
  6. Power

ANALOG AUDIO CONNECTIONS

The analog I/O connections on the ULN-2 have been engineered for maximum flexibility in that they support both balanced and unbalanced connections with a wide range of input and output levels and a wide range of matching impedances. This means that ULN-2 handles sources from mic level to line level and from mic impedance to guitar impedance. With that in mind, there are a number of aspects of the design that you should take into account when interfacing with ULN-2.

There are really three distinct analog input stages available in a ULN-2 input:

  1. The Mic amp, which is fed by the XLR portion of the Combo connector.
  2. The DI amp which is fed by the TRS portion of the combo connector.
  3. The TRS return jack. This is a line level input which is the shortest path to the A/D converter.

Each input path is optimized for specific sources, but each is capable of handling a wide variety of sources. For example, both the Mic amp and the DI amp are capable of receiving Line level inputs. Additionally the DI input is capable of 63 dB of gain and can be used with dynamic microphones (phantom power is only available with the Mic Amp).

Feel free to experiment with the different input paths and choose the one which works best for a given application.

Whenever possible, use balanced interconnects with ULN-2. The performance of balanced interconnects is much higher and much more resistant to noise interference and electrical (power) wiring problems. The expense of balanced interconnects is not substantially higher than unbalanced connections, so if the gear that you are interfacing with supports balanced connection — use it.

If you cannot utilize balanced interconnects, there are connection schemes that you can use that will maximize performance.

On input, at line level, it is sufficient to simply use standard unbalanced (TS) connections. If you are interfacing with the ULN-2 XLR inputs, you will need to ensure that pin 3 is grounded in the unbalanced adapter cable. The ULN-2 XLR inputs are all wired pin 2 hot and the 1/4” inputs are wired Tip hot.

TIP: To use the ULN-2 TRS input with guitar or bass, you can simply use a standard TS guitar cable (patch cord) and it will work fine. However, you can take advantage of the balanced input design of the ULN-2 to get more noise rejection than you thought possible on a guitar input.

In order to do this, you will need to make a psuedo-balanced telescoping shield guitar cable. This can be constructed with a TRS connector, a TS connector and balanced microphone cable. This cable will treat the guitar as a floating balanced source and provide a telescoping shield from the ULN-2 ground.

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Telescoping Shield Cable for Instruments

If you want to use the TRS inputs with balanced microphones, you will need an XLR female to 1/4” TRS balanced plug adapter cable. These are available commercially, or you can construct one easily. The connections are Tip to Pin 2, Ring to Pin 3 and Sleeve to Pin 1:

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XLR to Balanced TRS Cable

On output, the situation is a bit more complex. If you are driving an unbalanced load, you will get the best performance by not connecting the ring of the TRS jack to ground. In order to do this, you can simply use a balanced TRS/TRS connector with the unbalanced gear. You can also construct a special cable with a TRS connector and a TS connector. In this cable, you just let the ring of the TRS connector float:

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TRS to TS unbalanced cable

Alternatively, the TS connector can be replaced with an RCA connector for interfacing with gear that has RCA unbalanced interconnects.

MAKING THE 1/4” CONNECTION

When you connect a 1/4” plug to a ULN-2 jack, insert it straight and firmly, ensuring that the plug is fully inserted into the jack. If the plug is not fully inserted you will get level shifts, phase flips, distortion, or no sound.

To disconnect a 1/4” plug, firmly pull the plug straight out from the connector body. The connectors on ULN-2 are stiff, so you may have to exert some force to remove the plug.

MAKING THE XLR CONNECTION

When you connect a Male XLR plug to a ULN-2 jack, ensure that you have aligned the pins with the connector body and insert firmly until the retention tab clicks.

To disconnect the plug, press the metal retention tab flush against the box, and pull the plug from the ULN-2.

COPPER-BASED DIGITAL AUDIO

ULN-2 supports 2 channels of digital audio over copper-based connections. These connections can be made using either S/PDIF interconnects with the RCA connectors or with AES interconnects using the XLR connectors. Even though only one of the AES or S/PDIF inputs can be active at any given time, you can have different digital sources connected to each of the input connectors at the same time – you use the MIO Console application to select the active input. Audio routed to the digital outputs will be mirrored by both S/PDIF and AES outputs. This allows you to send the same stereo pair to two devices at once.

We recommend that you use the AES interconnect mechanism to establish the digital communication between the ULN-2 and other digital devices. The jitter and electrical noise tolerance on AES interconnects is substantially better than with S/PDIF interconnects. The AES interconnect standard is equivalent to balanced audio interconnections. If you need to use S/PDIF interconnects, try to use the shortest cables you can and, if possible, use special purpose 75 ohm S/PDIF or video cables.

The RCA connectors used for S/PDIF are friction fit coaxial connectors. When you connect them, ensure that they are fully inserted and tight.

The XLR connectors used for AES are fully locking. When connecting to them, make sure that you align the pins and insert firmly. When you remove the connector, make sure that you release the lock by pressing the lock release button before you pull the connector out of the ULN-2.

INTEGRATED SRC

Normally, when working with digital audio transport, you must take care to ensure that all devices communicating with one another are synchronized to the same audio clock. While this is still an important consideration with ULN-2, the hardware provides a special feature to simplify copper-based digital connections to the box. The digital input on ULN-2 has an optional asynchronous sample rate converter (SRC) that will automatically match the sample rate of the incoming audio to the sample rate of the ULN-2. This converter is enabled by default and you can disable it in the System section of the MIO Console. If you have synchronized the ULN-2 to the external source (using any of the extensive synchronization methods provided by ULN-2), you will generally want to disable the SRC in order to get 24-bit transparent audio transport over the digital input.

OPTICAL-BASED DIGITAL AUDIO

The ULN-2 Expanded provides two TOSLink™ connectors on the back panel. One is a transmit connector and the other is a receive connector. These connectors are used with Plastic Optical Fiber (TOSLink) cables to communicate with other devices. The TOSLINK connectors can be used to communicate with either the ADAT® Optical communication protocol or the Optical SPDIF communication protocol. Each port can be indpenedently switched between the two protocols via MIO Console.

The ADAT Optical standard allows a device to transmit 8 channels of 24-bit audio at up to 50kHz along with digital audio clock information.

The Optical SPDIF communication protocol allows a device to transmit 2 channels of 24 bit audio at 96kHz, along with digital audio clock information.

Since Mobile I/O provides direct routing within the box, you can easily configure the unit to work as an ADAT based 8 channel A/D/A. Refer to the chapter on MIO Console for information about configuring the routing.

CLOCK SYNC

Clock sync is a serious consideration in any digital audio system.

If you are recording analog sources with ULN-2, you can simply use the unit’s high-quality internal clock source to drive the converters. This is the easiest case to deal with.

If you need to interface with other devices digitally or ensure sample accurate sync with video sources, the extensive clock synchronization capabilities of ULN-2 will prove to be more reliable (and better sounding) than most higher priced alternatives.

There are four different ways to get external clock information into the unit:

  1. Sending a 1x word clock signal into the WC Input BNC.
  2. Sending a 256x word clock signal into the WC Input BNC.
  3. Sending an AES or S/PDIF signal into the Digital input.
  4. Sending an ADAT signal into the Optical Digital input.

The BNC word clock input port is a 75 Ohm terminated coaxial input. It should be driven by a 75 Ohm source driver and interconnected with 75 Ohm coaxial cable. If you do not use proper cabling and source drive, you will introduce reflections on the word clock cable which will propagate jitter into the recovered word clock. This is true whether you use the port as a 1x WC input or a 256x WC input, but becomes more important when the clock signal is 256x.

1x is generally appropriate for use with devices that provide a word clock output. If your device provides a 256x output, you may find that you get better results using that clock signal. The Digidesign® line of Pro Tools® products use 256x as their “ SuperClock™” clocking signal.

The AES recommended procedure for distributing clock is to use an AES clock signal. The AES clock signal is an AES digital audio signal with no audio activity. ULN-2 only uses the AES preambles for clock recovery, so it is immune to data dependent jitter effects. This means you can reliably use the Digital Input as a clock source with or without audio data.

FIREWIRE

FireWire® is Apple’s registered trademark for the IEEE 1394 High-Speed Serial Bus. FireWire started as an Apple technology to replace a variety of interface ports on the back of the computer. After promulgating a number of closed proprietary technologies in the early days of the Macintosh, Apple determined that open standards were better for the Mac, for the industry, and for Apple itself. On that basis they opened their technology for standardization under the auspices of the Institute of Electrical and Electronics Engineers, Inc. (IEEE), an international organization that promotes standards in the field of electronics. FireWire was standardized as IEEE 1394 and promoted for open licensing in the industry.

The first widespread adoption of the technology was for DV camcorders where space was at a premium and bus powering was not perceived as a real issue since all camcorders have batteries. Sony designed an alternative version of the standard 6-pin FireWire connector that provided 1394-based communication with 4-pins in a much smaller form-factor. This version of the connector sacrificed bus-power support and mechanical stability for reduced space requirements. Sony dubbed this version of IEEE 1394 “i.Link®.” This became the de facto standard in the DV world, and was later added to the IEEE 1394 standard. Both i.Link and FireWire refer to the same underlying standard and are completely interoperable. Obviously, i.Link connectors and FireWire connectors cannot be used together without adapters.

ULN-2 uses the FireWire flavor of the IEEE1394 connector with 6-pins for bus power support. The unit ships with two 6-pin to 6-pin FireWire cables, one that is 0.5 meters long (about 18 inches), and the other 4.5m (about 14.5 feet) long. If you want to use ULN-2 with a 4-pin FireWire device, you will need to purchase a 6-pin to 4-pin adapter cable. These cables are available from a wide variety of retail sources. If you are using a 4-pin cable to connect any device to the computer with ULN-2, bus power will not be available.

The 6-pin FireWire connector is polarized by its shape, one end of the connector is pointed. The FireWire ports on ULN-2 point downwards toward the bottom of the box. It will be very difficult to insert the connector upside down, but it is possible if you force it. If the plug is inserted into the socket upside down, the socket will be destroyed. NEVER FORCE A FIREWIRE CONNECTOR INTO A FIREWIRE SOCKET.

Devices connected to the FireWire bus are autoconfiguring. You do not need to set IDs or DIP switches or in any way configure the devices in order to facilitate communication between devices or to configure the bus.

FireWire devices on the same bus must be connected in a tree structure with no loops. This means that devices can be connected to each other in any order, and any device with multiple ports can act as a chain or a hub for other FireWire devices, but you should never be able to get from one device to another by more than one path. If you construct a loop in the bus, it will not operate properly and you will not be able access some or all of the devices on the bus.

Although you are able to attach devices in any order on the FireWire bus, the order of attachment will have an impact on performance. Most current model FireWire devices support 400 Mbs operation, but many older devices may only support 100 or 200 Mbs operation. These devices act as a bottleneck in the bus and limit the speed of any bus traffic that flows through them. In order to maximize performance, you want to ensure that low speed devices are not used to join high speed devices. In practice this generally means that you should attach your ULN-2 directly to your computer or through a high speed hub.

To connect ULN-2 to your computer simply plug a FireWire cable into the ULN-2 and into the computer. The FireWire bus provides a path for all communication between the computer and ULN-2 – audio, control and meter data.

ULN-2 audio transport takes advantage of FireWire’s support for isochronous transmission, in which the ULN-2 can reserve a dedicated amount of bandwidth on the bus for moving audio samples. Since the audio must be transmitted on a regular basis to ensure continuous playback and recording , the isochronous model is perfect.

Control changes and meter data are transmitted using asynchronous transactions on the FireWire bus. This transmission approach makes use of the unreserved bandwidth on the bus and competes with things like FireWire hard disk accesses for time. Under normal circumstances this is completely transparent to the user. If the bus becomes overloaded, you may find that disk accesses and meter updates slow down. If you are experiencing bus overloads, you can always add a second FireWire bus with a third-party FireWire card (PC-Card or PCI card depending on your machine), and offload one or more devices to the second bus.

POWER

One of ULN-2’s many great strengths is the flexibility of its power system. ULN-2 can be powered from any DC source (including bus power) in the range of 9V to 30V as long as it provides 12 Watts of power. The DC inputs on ULN-2 are a 2.1mm coaxial power connector, center positive and a 4-pin XLR connector Pin 4 Hot. So if you are powering the unit with a third party power source and it supplies 9V, the power source will have to provide 1.4 amps of current. If you are powering the unit with 12V, the power source will have to provide 1 amp of current, and so on.

The ULN-2 ships with a world-ready 24 volt, 2 amp power supply. You can plug this supply into any AC power source from 90V to 240V, 50Hz - 60Hz, using an appropriate IEC power cord, and it will supply the proper power to the ULN-2 on the 2.1mm coaxial power connector. ULN-2 will automatically supply the extra power to the FireWire bus. This means that the ULN-2 and its power supply can be used to power other bus-powerable FireWire devices including hard-drives, hubs, and other ULN-2 units.

Since ULN-2 is DC powered, you can also power up the ULN-2 using the FireWire bus or another DC source. The ULN-2 uses 12 Watts of power, so the device supplying the bus power must be capable of sourcing that much power. Most desktop Macs provide more than enough power for ULN-2 and one other low power device. Most laptops provide enough power for ULN-2, but not enough for ULN-2 and another bus-powered device at the same time. If you are using a Powerbook computer, you should not expect to be able to power both the ULN-2 and a hard drive from the computer. The power capabilities of individual computers vary, so you will have to test the complete system to determine exactly how much your computer can handle.

If you find that the computer is not capable of powering ULN-2 or does not provide enough run time, you may want to explore using an external power source with the ULN-2. Check with Metric Halo for details on different battery power solutions for ULN-2.

As with all electronic devices, when connecting an external power source to the ULN-2, you should first connect the power source to ULN-2 while it is in an unenergized state (e.g. not connected to the mains or switched off). After the connection to ULN-2 has been made, you should energize the power source.

If you connect an energized power source to the ULN-2’s 2.1mm power connector you may see a small spark when you make the connection. This is due to surge current and is normal if you connect a power source in this way. While this will not damage the ULN-2 in any way, to avoid the spark just connect the power connector to ULN-2 before connecting the power source to the wall.

 ULN-2 Overview

ULN-2 Overview

Thank you for purchasing a Mobile I/O ULN-2™, the ultimate FireWire®–based professional audio interface. Your ULN-2 provides an array of functions that allow you to record and mix with unprecedented quality – Anywhere, Anytime.

What it is

ULN-2 is the result of a dream to create a piece of audio gear that provides unbelievable audio quality while at the same time offering a degree of mobility and convenience that until very recently was simply not possible. The successful integration of world-class analog stages, excellent A/D/A conversion, and the amazing digital mixing, routing and FireWire connectivity that has already made the Mobile I/O line famous, places the ULN-2 in a unique position among computer audio interfaces.

ULN-2 is a portable, high–quality, modular FireWire–based multi-format audio converter, interface, and processor for professional audio applications. The ULN-2 is equipped with two balanced analog inputs on Neutrik™ combo connectors, two channels of Digital I/O (AES/EBU and S/PDIF), two balanced analog outputs (1/4" TRS), two balanced monitor outputs for connecting directly to power amps and self powered monitors, as well as wordclock in/out and 2 IEEE 1394 FireWire connectors that support 400 Mbs operation. All inputs and outputs are capable of 24-bit/ 96kHz operation.

What it has

What you need to use it

What comes with it

Your ULN-2 package contains the following items: If any of these items are missing from your package when you open it, please contact Metric Halo or your dealer immediately for assistance.

 v.5 Mixer Overview

Mobile I/O v.5 Mixer Overview

This document provides an overview of the v.5 Mixer Architecture with configuration and operation quick-start guide.

Ultra-Quick Start Guide

Perhaps the most fundamental thing for you to understand is that the v.5 mixer model is based upon user-configuration. The advantage to this is that you can build the exact mixer you need for any specific task or set of circumstances. But it does mean that the v.5 mixer window is blank until you configure it.

In order to avoid the “blank page” syndrome you can use one of the templates we've included with the software. The template selection dialog is displayed on first launch, and it can be accessed at any time from the File menu.


Template Selection Dialog

Alternatively, you can use the following steps to configure a mixer from scratch.

The easiest way to create a new mixer configuration is to use the Mixer » Configure Mixer… menu command. Selecting this command presents the Configure Mixer sheet, which allows you to do bulk configuration of the mixer.


Configure Mixer Sheet


The sheet is split into two major sections:

You create and configure the mix busses using the top section, then you configure the channels you assign to the busses in the bottom area. When you select a mix bus from the table in the Mix Busses section, the Selected Mix Bus Configuration area automatically updates to reflect the current configuration of the selected mix bus. All of the available input channels are listed in the Selected Mix Bus Configuration, and you can assign any of the channels to the selected mix bus by checking the channel’s Enable check-box.

To Create a Mix Bus:
  1. Click the “+” button — this creates a new stereo bus with a default name.
  2. Double-click the new bus in the list; you can now edit the bus name.
  3. You can adjust the bus type with the popup in the Bus Type column.
To Delete a Mix Bus:
  1. Select the bus you want to delete
  2. Click the “-” button — this deletes the bus and de-assigns the channels assigned to the bus (but it won’t delete the channel strips from the Mixer window).
To Add an input channel to a Mix Bus:
  1. Select the bus to which you want to add a channel.
  2. Check the box in the “Enable” column for the channel you want to add to the bus.
To Remove an input channel from a Mix Bus:
  1. Select the bus from which you want to remove the channel.
  2. Uncheck the box in the “Enable” column for the channel you want to remove from the bus.

When you have finished configuring your mixer, click the Configure button in the bottom right. The mixer you have configured will now appear in the Mixer window.

The v.5 Mixer removes the direct-route connection between the physical inputs and FireWire that has existed in the Mobile I/O since it was originally shipped. Instead, the v.5 Mixer extends and enhances the concept of FireWire returns. In fact, all audio is now sent to the computer via FireWire returns. In other words, if you want to send audio from your Mobile I/O's inputs to your computer, you will need to assign those inputs to FireWire returns manually via the Direct Outs in the mixer input strips. Hint: selecting “Auto” in an input strip's direct output pop-up menu will automatically assign the direct out to the next available FireWire output.

The Mixer you have created will automatically contain Input channel strips for all of your configured channels, and a Master channel strip for the Mix Bus. Now you'll just need to route your Master strip to the desired physical outputs using the Bus Output pop-up menu at the bottom of the Master strip, and you can hear all the inputs assigned to that mix bus through your MIO outputs. See below for more details about Master Strips.

Ultra-Quick Start Summary

To quickly configure a mixer that will send your selected inputs to your studio monitors and to your computer:

  1. Select the Mixer » Configure Mixer… menu command.
  2. Create a Mix Bus as described in “To Create a Mix Bus” above.
  3. Add input channels as described in “To Add an input channel to a Mix Bus” above. Be sure to add some physical inputs and some DAW inputs so you can route both physical and computer inputs through the mixer.
  4. When you have finished Enabling the input channels you wish in your mixer, click the Configure button in the Configure Mixer sheet to create your mixer.
  5. In the Input channel strips for each of your physical inputs, click on the Direct Output pop-up menu (either pre- or post-send, up to you), select "Auto." This assigns each Input strip to a FireWire output.
  6. In the Master Strip, click the Bus Output pop-up menu and select the physical outputs to which your studio monitors are connected.

That's it! You've just created your first v.5 mixer. You can now route the audio signal from your Mobile I/O's physical inputs to your computer, from your computer to your Mobile I/O, and connected the mixer to your studio monitors.

Introduction

The v.5 Mixer Architecture represents a quantum leap forward in both the power and usability of the Mobile I/O's mixing/routing/signal processing engine. This new architecture builds upon signal processing elements that have been in development at Metric Halo since the initial introduction of the Mobile I/O in 2001.

With the introduction of the v.5 architecture, we have taken the basic interaction model of MIOConsole and we have re-engineered it based on the feedback we've received from our users and beta-testers over the years.

The v.4 took the approach of defining precisely what the hardware was capable of; you had to adjust your workflow to fit its capabilities. The v.5 mixer takes the approach of assuming nothing about the configuration of the hardware, allowing you to define your configuration up to the limits of the hardware. This has a number of significant benefits to the user:

But again, keep in mind that the consequence of the v.5 model being based upon user-configuration is that the Mobile I/O is a blank slate until it has been configured by you.

Routing

The v.5 Mixer removes the direct-route connection between the physical inputs and FireWire that has existed in the Mobile I/O since it was originally shipped. Instead, the v.5 Mixer extends and enhances the concept of FireWire returns. In fact, all audio is now sent to the computer via FireWire returns. The following illustrates the overall routing structure of the Mobile I/O with the new v.5 mixer:


v. 5 2d Routing Model

As you can see, instead of the hardwired direct-routing of inputs found in the previous Mobile I/O console, now all of the audio streams that are sent to the computer have been converted to FireWire returns. The mixer now provides the facilities to easily route any of the audio in the system to the computer — even between applications!

Routing is critical!

It is very important to understand that the v.5 mixer does not provide default routings!

You must choose where you want your audio to go. If you don't route a mix bus to a physical output, you will not hear any audio. If you don't route an input to FireWire it will not be available to any applications in your computer.

Routing in v.5 is really easy — so just remember — if you are not hearing or seeing your audio where you expect to, check your routings!


A Quick Tour of the v. 5 Mixer


The v.5 mixer provides many new features as well as greatly simplifying the use of older features. The mixer provides the following functions:

Thanks to the new routing functionality in the mixer, we have been able to completely remove the patchbay from the system. With the new routing and plugin functionality in the mixer, coupled with the availability of the Graph plug-ins, we have also been able to completely remove the overall +DSP graph interface from the system. These changes make utilizing the power of the Mobile I/O and +DSP far more elegant and intuitive.

The Mixer, Routing and +DSP panes in the MIO Console window are no longer required when all the units you are using have 2d Cards installed. If your entire system is 2d Card enhanced, you can disable these panes in MIO Console preferences. If you continue to use units that do not have 2d Cards installed, you will still use those panes in the MIO Console window to control those elements of your legacy units.

Technology

The v.5 mixer is based upon the +DSP graph technology originally designed for +DSP. With the 2d Card, we have created the concept of an Übergraph. The Übergraph is an internal +DSP graph that represents the entirety of the Mobile I/O, including the 2d DSP, all the physical I/O and the FireWire I/O. Since the +DSP graph provides infinite routing and multing capabilities with integrated latency compensation, the addition of full access to all I/O resources means that everything can go everywhere in the new system.

What’s up with the Übergraph?

While the v.5 mixer is based upon the Übergraph, it does not require (and, in fact, does not allow you) to interact directly with the Übergraph. Rather, the v.5 mixer manages all of the plug-ins and connections within the Übergraph for you. So, even though you may be creating an incredibly complex network of plug-ins with many mults and bus to bus routes, the user interaction required for you to control the system is simple and direct. This explanation of the Übergraph is included to give you a deep understanding of the v.5 architecture, nothing more.


The v.5 mixer works as complex routing manager, adding plug-ins (including mixer plug-ins) to the Übergraph based upon your configuration commands. Because the +DSP engine automatically configures the runtime environment to compensate for routing latency in the configured DSP graph, you can rely on the fact that bus to bus routes all arrive with no routing latency. This allows you to configure sub-mixes and stems, including sub-bus processing, without having to worry about phase-cancellation problems or other latency related issues.

The v.5 mixer builds a model of your desired mix bus structure and also tracks all of the routes that you create between busses, bus outputs and direct outs. The mixer then ensures that everything stays routed properly as you add and remove plug-ins, channels and busses. This technology makes the manipulation of your mixer and routing simple — rather than think about how to route the elements of the mixer, you simply insert things where you want them and the mixer takes care of all the routing for you. Since the model is based upon the +DSP graph, you have much more flexibility in your routing options with regards to arbitrary sends, I/O, direct outs, and mults than you have in most digital mixers.

Configuration

As described above, the v.5 mixer starts with a clean slate — nothing connected:


Blank Mixer Window

In order to use the Mobile I/O with v.5, you must have a mixer configured. This means that you need to load one of the supplied mixer templates (using the "Open Template..." command), a saved MIO Console Settings file (if you have one), or you can create a new mixer configuration.

Using Templates for Configuration

The easiest way to create a new mixer configuration is to start with a template using the File » Open Template… menu command. When you select this command, you are presented with the "Choose Configuration Template..." window:


Template Window

To start with a template, click the template you would like to use in the "Template" list on the left side of the window. A description of the template will appear in the "Template Description" area to give you more information about the template. You can browse through the templates to choose which one is the best starting point for you. For example, if you select "2882 Basic Setup", you will see the following description:

Selected Template

Once you decide to use the selected template, click the Open button. If you decide that you don’t want to use one of the templates, you can click the “Use Current State” button to use the state that is currently loaded in the console. If you are using the Template dialog at application launch, this will load the state that the console was in the last time you quit MIO Console. If this is your very first launch ever of the MIO Console, you will then need to manually configure your mixer.

Manual Configuration

The next way to create a new mixer configuration is to use the Mixer » Configure Mixer… menu command. When you select this command, you will see the Configure Mixer sheet:


Configure Mixer Sheet

The Configure Mixer Sheet allows you to do bulk configuration of the mixer. The sheet is split into two major sections:

You create and configure the mix busses using the top section, then you configure the channels you assign to the busses in the bottom area. When you select a mix bus from the table in the Mix Busses section, the Selected Mix Bus Configuration area automatically updates to reflect the current configuration of the selected mix bus. All of the available input channels are listed in the Selected Mix Bus Configuration, and you can assign any of the channels to the selected mix bus by checking the channel’s Enable check-box.

Each mix bus has a number of attributes that you can control:

The Bus Name is the name you assign to the bus. It is used throughout the Mixer UI to identify the bus. You can name the bus anything you like.

The Bus Type determines the number of channels of the bus (mono, stereo, etc.). The type of bus determines the type of panner used to connect the input-strips to the bus. We have implemented the following types so far:

The Bus Mode allows you to determine if the bus has a master fader or not. If the mode is Master Mix the mixer will create a master fader for the bus. If the mode is Aux Bus the mixer will not create a master fader, but you can assign the bus to another bus to create a return fader.

Mixer Configuration Tasks

The following task lists are succinct guides to performing the specific configuration tasks you will need to engage in while configuring a mixer using the Configure Mixer Sheet.

To Create a Mix Bus:
  1. Click the “+” button — this creates a new stereo bus with a default name.
  2. Double-click the new bus in the list; you can now edit the bus name.
  3. You can adjust the bus type from the pop-up menuin the Bus Type column.
To Delete a Mix Bus:
  1. Select the bus you want to delete
  2. Click the “-” button — this deletes the bus and de-assigns the channels that were assigned to the bus (but it won’t delete the input channel strips in the mixer).
To Add a channel to a Mix Bus:
  1. Select the bus to which you want to add the channel.
  2. Check the box in the “Enable” column for the input channel you want to add to the bus. Your choice of input channels consists of any of the physical analog and digital inputs from your Mobile I/O, as well as any available inputs from your computer (called "DAW" inputs). The channels you select here are the channels that will be available in your mixer as Input strip destinations.
To Rename an Input Channel in a Mix Bus:
  1. In the "Channel Name" column, double-click the name you wish to rename.
  2. Type the new name of the input channel.
To Remove a channel from a Mix Bus:
  1. Select the bus from which you want to remove the channel.
  2. Uncheck the box in the “Enable” column for the channel you want to remove from the bus.
To Add/Remove all visible channels to/from a Mix Bus:
  1. Select the bus you want to configure.
  2. <option>-click one of the checkboxes in the “Enable” column. All visible channels will be added or removed from the Mix Bus.
To limit the channels visible in the Selected Mix Bus Configuration area:
  1. Select the bus you want to manipulate.
  2. Type the text that you want to use to limit the channels visible in the table.
    • You can click the pop-up menu next to the search field to quickly enter certain standard search text items (like “Analog” or “DAW”)
    • You can click the pop-up menu inside the search field to choose which column is searched for the text
  3. To remove the limitation, delete the text in the search field or click the X in the circle in the search field.

Input Strip Details

Every Input strip in the v.5 mixer has a similar set of controls. The following figure shows each element with a label; a detailed description follows below:


v.5 Mixer Input Strip

Selection-based Linking

When you have multiple strips selected, the changes you make to a selected strip will be applied to all the selected strips. For example, if you have multiple input channel strips selected and you choose a setting for "Character" on one of the selected strips, that setting for Character is applied to all selected strips.

This feature allows you to quickly apply bulk changes to the state of the mixer.

The parameters that are linked by selection are: The following parameters are linked by selection, but the link is only applied if the <control> key is held down when changing the parameter:

Plug Ins

The v.5 Mixer provides an insert model for using +DSP plug-ins. The 2d Hardware includes a basic set of plug-ins that provide "nuts & bolts" production processing:

When the +DSP license is added to 2d, the options grow dramatically. All +DSP plug-ins can be inserted directly into the insert slots in the channel strips, or you can insert a +DSP graph into any of the insert slots, and then insert and connect a graph populated with +DSP plug-ins within the inserted graph.

The plug-ins that may be inserted in any given slot depend on the number of channels of a given input channel strip. All mono plug-ins may always be inserted in any slot; if you insert a mono plugin into a strip that has a multichannel input (for example a bus master strip or bus return strip, or a multichannel input strip), the mixer will automatically instantiate multiple copies of the plugin (for example — two plug-ins into a stereo strip and 5 plug-ins into a 5.0 strip), and link the parameters of the instances so that when you control the inserted plugin, it will control all instances.

If you are working with a multi-channel strip, only plug-ins that make sense for the number of channels of the strip will also be available; for example, with a stereo strip, you will see both the mono and stereo versions of the MIOComp and MIOLimit dynamics processors; you can select the version that works best for you. At the present time, there are few processors that have specifically been built for multi-channel strips with more than two channels. If you expect to use (or change to) a bus with multiple channels beyond stereo, you will probably want to use the mono version of the plugin as they can be automatically instantiated as you adjust the number of channels of the strip.

Plug In Graphs

When you insert a graph in the mixer, the graph is automatically generated with input and output ports to match the number of channels of the strip that it is inserted into. The default state of an inserted graph is for the inputs to be connected directly to the outputs.

When you open the graph UI for the insert, you can insert any set of plug-ins into the graph that is shown in the graph UI window. These plug-ins can be connected by virtual cables, their UIs opened, and parameters set. The graph I/O connections will automatically be routed to the appropriate points in the strip that hosts the graph. The graph will be saved and recalled with the rest of the mixer, and you can also choose to save the graph independently as kind of a "macro" that can be inserted again and again into the mixer.

The graph inserts also have access to the saved graph patch library — this lets you migrate patches that you have created in +DSP in v.4 or earlier to inserts in v.5. You can also save the configuration of your graph insert into the library as a preset, so as you work with the mixer, you can build up a set of "secret weapon" processors that can be instantly recalled and tweaked as needed.

To Migrate Graph Patches from v.4 or earlier Consoles:

Graph patches created in v.4 or earlier consoles may require a little clean up for use with v. 5.

• If the graph patch you want to load is already set up to take input from Analog 1 (for a mono patch), Analog 1-2 (for a stereo patch) or Analog 1-N (for a multi-channel patch), then you can just use it directly in v.5.

• If the graph patch you want to migrate is set up to take input from another source (say Analog 6, or Digital, or DAW, or ADAT), you need to open the graph in the Virtual DSP area of the +DSP pane and change the inputs to patch to Analog 1 (for a mono patch), Analog 1-2 (for a stereo patch) or Analog 1-N (for a multi-channel patch). The same applies for output assignments. Make sure to set the outputs to Process Bus 1-N. Save the patch with the new input and output assignments and it is ready to be used in a v. 5 graph plug in.

Also, please note that v. 4 and earlier medium and long delays will not currently load on the 2d DSP. However, MIO Console v.5 includes new 2d compatible delays that you should be able to use to replace the older medium and long delays.


Graphs may be instantiated into any insert point of any type of channel strip, whether it is an input strip, aux return strip or a master fader strip. This allows you to configure (or utilize pre-configured) processing in a variety of ways as appropriate for the project that you are working on. For example, you can insert a one in-one out graph into a mono input strip to implement anything from a guitar amp model to a parallel compressor or a feedback delay. You might insert a 2-in 2-out on an aux return strip and build a stereo reverb or echo time domain processor to process the send bus; you would insert sends on the various input strips to route the audio to your reverb graph.

Plugin Macros

With a +DSP license, the v.5 Mixer also supports the direct insertion of Plugin Macros, which are premade Graphs. Metric Halo includes a number of bonus macros with the +DSP license, including reverbs, guitar processing models, mastering tools, delays and other effects. Some of the macros are open — once you insert the macro, you will have full access to the graph, and you can edit it, modify it and interact with it as you please. Other macros are closed and represent a monolithic processor; you can insert them, and they do the job they were designed for, but they cannot be edited.

Sends

Any insert in the mixer may be used to send the audio from that point in the signal path to any bus defined in the mixer. When you insert a send to a bus on a strip, a new send strip is automatically added to the mixer; when you insert a send or click on a send tile in the mixer, MIOConsole will automatically open the Sends Mixer window:


Sends Mixer Window

The Sends Mixer window shows the all the send strips for the currently selected bus. Send strips are more limited than full input strips. Each Send Strip has a panner (if appropriate for the bus width and send width), phase invert, solo, mute and pre/post fader buttons, as well as a send level fader and level meter. All of the elements common with the full input strips function in the same way as the elements in the full strip.

The control element that is unique to the send strip is the Pre/Post Fader button (the lavender button with the "P" in the illustration above). When this button is illuminated, the send functions "Post Fader" relative to the mixer fader for the strip that the send is inserted on. This means that the level of the signal will be adjusted by the input channel strip's level fader. This is default state of the Pre/Post Fader button. Since the signal is actually routed from wherever the send is inserted, the "Post Fader" state does not control the routing of the send, but rather the level; the total send level will be the sum of the source fader level and the send level fader. When you have selected "Post Fader" mode for the send, the send will also respect the mute and solo state of the strip that the send is inserted on.

In "Pre-fader" mode, only the send level fader controls the level of the signal to the send destination. Also, the send does not take the mute or solo state of the strip into account when sending the signal to the send destination.

Master Strips

Master Strips are used to control the signal processing and output routing for mix busses. Rather like the Input Strips, they provide Character, Direct Outs, Inserts, and Mute controls. They also provide a very flexible strip output routing popup that allows you to route the output of the processed mix bus to any specific output path, mult it to many output paths, or to assign it to the Monitor Controller.


v.5 Mixer Routing Connections

Routing Summary

All routing in v.5 is managed through the mixer and monitor controller. The diagram below summarizes the routing control points and what elements of the routing model they control:


v.5 Mixer Routing Connections

In the diagram above, the various colored arrows indicate different routing elements:

These routing elements allow you to completely configure the routing resources of the mobile I/O and accomplish any set of routings required quickly and easily, directly from the signal source.