MIDI is, to some, a great complicated mystery that they may never wrap their minds around completely.
However, when understood even in the most basic sense, MIDI can be harnessed and used to make the impossible possible.
I’ve designed this article to basically get you in touch with the basic concepts of the MIDI universe.
First, MIDI is an acronym which stands for Musical Instrument Digital Interface.
It is a digital information protocol developed in 1983. Being a digital information protocol it basically means that MIDI is only 1s and 0s. It is a language that allows communication between musical instruments and related devices that are MIDI capable (sequencers, computers, sound modules, samplers, etc.)
MIDI contains NO actual audio information. It is a digital communication protocol which contains only numerical commands.
Initially, many companies became interested in MIDI and its promises of grandeur, but this in turn created problems, which were remedied through the introduction of General MIDI.
General MIDI is a STANDARD protocol that was implemented in 1993. It is a set of specifications created for the soul purpose of easy communication between all MIDI equipment, regardless of the maker. You might think of it as a sort of universal language of electronic devices.
These are the basics of a MIDI signal:
Note ON – Tells the device when it should begin the note.
Note Number – Indicates the note which should be played. (Each and every note that can be played by musical instruments has an assigned number)
Note Velocity – Determines the acoustic intensity or volume of the note to be played. This can be a value anywhere from 0 to 127—0 being the quietest and 127 being the loudest.
Note OFF – Indicates that the note should stop being played.
This is not the only information included in the signal, but the most important. Additional Information contained in a MIDI signal includes timing information, pitch bend, program changes, channel aftertouch, polyphonic key pressure, sustain pedal, running status, etc.
A MIDI channel is basically used to send each individual part of your MIDI composition to its own individual place in your MIDI network. Without MIDI channels you could write a whole orchestral arrangement and have it placed through one single MIDI module with a piano patch loaded into its memory. To say the least, this would not be pretty.
Channel 10 will always be a percussion track. And, MIDI is capable of 16 channels.
MIDI modes are the different modes in which a MIDI device may operate. They include:
(MODE 1) OMNI ON – POLYPHONY
(MODE 2) OMNI ON – MONO
(MODE 3) OMNI OFF – POLYPHONY
(MODE 4) OMNI OFF – MONO
What does that mean? When in OMNI ON mode, the MIDI device allows the unit to respond to all incoming data regardless of its channel. When in OMNI OFF mode, the MIDI device will respond to incoming data on only one specified channel.
When POLYPHONY is engaged, the MIDI device will allow multiple notes to sound at the same time.When Mono is engaged, The MIDI device will allow only ONE note to sound at a time.
What are they and what do they do?
MIDI IN – Allows for the input of MIDI data; in other words, it is the RECEIVING port.
MIDI OUT – Allows for the output of MIDI data; in other words, it is the SENDING port.
MIDI THRU – This one’s different. Simply put, it DUPLICATES the MIDI data going into the MIDI IN port and sends it out.
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.
The pros and cons of transforming your analog signals into a digital stream
In the world of networked audio transport, there are two major categories in which a system may fall: a fully standards-based network, or a proprietary network that may or may not use standards-based transport.
Both have their advantages and disadvantages and, of course, are subject in varying degrees to the problems associated with transforming an analog signal into a digital stream and then back again.
Let’s first explore the biggest question that a system designer should ask when someone shows them a new piece of digital gear: “What is the latency as it relates to audio being transported in large networks?”
Latency, or what some manufacturers call “propagation delay,” is the amount of time that an audio signal is delayed due to digital processes including analog to digital conversions (A/D), digital to analog conversions (D/A), and digital signal processing (DSP).
For live sound applications, excessive amounts of latency can wreak havoc on the audience as well as the performers by creating listener fatigue and poorly reconstructed audio.
Generally speaking, the extent to which latency will cause a problem is a function of the ratio between the direct sound and the sound that is delayed.
In a large-scale sound system, where there is little to no direct sound, the delay does not become a problem until there is a noticeable delay between sight and sound.
Pick your audio network transport highway: good ol’ CAT5 and up-and-coming fiber optic.
However, when the direct sound is within 8 dB or so of the delayed signal, 20 milliseconds (ms) to 30 ms difference will be audible. As for performers on stage, the acceptable time is generally shorter, especially when it comes to their monitors.
Echo from a system - whether it is acoustic or electronic - contributes to performer and audience fatigue.
A performer using an in-ear personal monitoring system will be conscious of any latency in the system to as little as 5 ms to 10 ms.
Small latencies on the order of 100 microseconds (µs) can cause phasing problems that can result in high frequency roll-off when they are added back into the mix with an un-delayed source.
A Common Clock
It should be noted that all digital network components should be synched to a common clock.
The best way to achieve this is with the use of a separate word clock.
This will allow all devices in the chain to be locked to each other and thus eliminate any phase shifts that may occur between devices from un-synched sampling rates.
With advances in digital technology, latencies for audio gear have dropped dramatically, to the point that some digital mixers are below 3 ms for analog in to analog out with no added time for DSP.
The problem lies in cascading multiple digital devices together if not digitally linked. When having to make A/D and D/A conversions for every piece of digital gear in a signal chain, it’s easy to see how the latencies from conversion alone can add up to an unacceptable total.
Two key factors in comparing digital audio networks are the sampling rate and the bit resolution that the network supports.
The sampling rate is the ”rate” at which a digital device ”samples” the composite analog sound waveform over time.
The sampling rate is important for the way digital audio can describe the frequencies in a sound.
Bit depth, or resolution, describes the potential accuracy of a particular piece of hardware or software that processes audio data. In general, the more bits that are available, the more accurate the resulting output from the data being processed.
For example, audio recorded with a 48 kHz sampling rate and 24 bits of resolution will have 48,000 measurements of which there are 16.7 million different values that each measurement can be per second.
Sampling rates and bit resolutions are important to digital audio networks because they need to remain consistent to keep the latency to a minimum.
A sampling rate conversion typically takes the same amount of time as an A/D conversion.
The major limitation on the number of channels a network can support is bandwidth, which is the amount of information that can be sent down the chain at one time.
The bandwidth required to pass one channel of audio varies with the sampling rate and the bit depth selected. As both increase, so too does the bandwidth demand reducing the maximum number of channels which can be transported over a particular network’s architecture.
Well, what if you could have just one A/D conversion at the start of the signal chain and one D/A at the end with all the other gear still in the chain?
Now we’re talking about the digital audio networks. Again, there are two groups of these digital audio networks, so let’s take a closer look at both.
A fully standards-based network abides by a published set of rules, which specifies a recommended interface for serial digital transmissions. This includes the description of data format for transport and the method of transport.
The best part of a standards-based audio network is that any manufacturer can implement the format into their products without having to pay the often-expensive licensing fees of a proprietary solution.
This allows the use of “manufacturer A’s” reverb unit with “manufacturer B’s” digital console without the conversion to analog, resulting in a total system latency that is much lower than if a D/A and A/D conversion was necessary.
The most common standard is AES3, more often called the AES/EBU (Audio Engineering Society/European Broadcasting Union) standard. AES3 uses a 110-ohm shielded twisted pair cable to send two channels up to 300 feet. The standard allows up to a 24-bit resolution with no maximum sample rate.
Another standard developed by the AES is AES10, more commonly called MADI (Multichannel Audio Digital Interface).
MADI offers 64 channels at a 48 kHz sample rate and 32 channels at a 96 kHz sample rate with a resolution of up to 24 bits per channel. Transmission over a single 75-ohm coaxial cable has a limitation of 150 feet, but the use of fiber-optic cables can extend the length to two miles.
Other multichannel standards include ADAT Optical (ADI), Sony/Phillips Digital Interface (S/PDIF), and Tascam Digital Interface (TDIF). See Table 1.
The only latency added into standards-based digital audio distribution is the act of making the A/D and D/A conversions and any subsequent DSP that occurs. This is unlike some proprietary solutions that also require additional time to transcode and transport the data.
Viable and Affordable
There are a multitude of proprietary systems in the ever-evolving marketplace which use the standard IEEE 802.3 Ethernet protocols.
Ethernet is used because it’s relatively inexpensive, reliable, and the technology has been, and will continue to be, upgraded by the computer industry.
There was a time when 10baseT was the best thing going. Now, gigabit (100baseT) Ethernet has become viable and affordable.
Table 1: A point of format comparison.
The problem with Ethernet is that it is a non-deterministic system, meaning that the data will arrive when it feels like it.
However, several manufacturers have developed systems that make the arrival times very predictable and allow for the network to be synchronized with only a small margin of error.
For instance, one standard’s master unit regularly broadcasts beat packets onto the network either from its internal clock or an external master clock.
Other devices on the network which utilize said standard lock onto the arrival time of this packet and regenerate the clock locally. The error in clock delivery is ±1/4 sample period; this translates to about .005 ms at a 48 kHz sampling rate.
While digital seems to be the future of audio networks, it still requires attention to detail and proper setup.
Monitoring overall system latency and keeping a consistent sample rate and bit resolution are both new requirements for the digital age.
The answer to the question of what system is best for your situation is still the one that best meets your needs, which might be good old-fashioned analog.
Digital is just another hammer in the sound designer’s tool belt which can either drive the nail home if used correctly or smash a thumb if its requirements are ignored. David A. McNell, CTS, is an AV engineer working in the Special Technologies Group of Newcomb & Boyd, Atlanta. He is a Certified Technology Specialist (CTS) and an EIT.
Assessing the CL5 and companion I/O units in the shop and in the field
The CL5 is the largest model in the new Yamaha CL Series of digital consoles.
The line-up comprises three consoles, the CL1, CL3 and CL5, all founded on the company’s Centralogic interface and ranging in scale from 48 to 72 mono plus eight stereo inputs.
All offer 16 DCAs, 24 mix/8 matrix output buses, eight mute groups, 300 scene memories, and recording options.
The CL5 provides three banks of eight channel faders in addition to eight Centralogic and two master faders, as well as an onboard meter section. The CL3 sports two banks of channel faders (plus eight Centralogic), while the CL1 has a single bank of eight faders (plus eight Centralogic) and both smaller consoles, with the meter bridge optional for both.
CL Series consoles can act as stand-alone units and feature on board inputs and outputs as well as the ability to utilize up to three mini-YGDAI I/O cards to expand the number of inputs and outputs, and they also work in tandem with the new RIO rack mountable I/O units.
The RIO3224-D offers 32 inputs and 16 analog outputs plus four stereo AES outputs, and the RIO1608-D features 16 inputs and eight outputs. The consoles connect to the remote stage boxes via Cat-5e or Cat-6 cable and run on Audinate’s scalable Dante network.
The Yamaha CL5 in the author’s shop. (click to enlarge)
While it’s largest of the three consoles, the CL5 is rather compact for the amount of features it packs. Dimensions are 41.5 x 26.25 x 11.75 inches (w x d x h), and weight is just under 80 pounds.
The top surface of the CL5 has two angles – a flat section where the faders and most controls are located, and an angled back section that houses the screen, meters, USB port, as well as the user defined knobs and the selected channel controls.
The large color touch screen is located in the center of the angled section and the selected channel controls for gain, EQ, aux sends, etc are located directly to the left of the screen.
Four user defined knobs are located directly to the right of the screen. The meters are located to the right. To the left is a large area that can hold an iPad or iPod and features a shelf to keep the phone or tablet from sliding down.
A look at the clean layout of the right side control surface. (click to enlarge)
The flat control surface features a Centralogic control section in the center that offers eight faders, each with an ON, CUE and SELECT button, meter and rotary knob. Nine function buttons to the right of the faders allow the operator to select between inputs, DCA, Mix, Stereo, Matrix or Custom layers.
The scene memory section and 16 user-defined keys complete the center section. To the left are two channel banks each with fader, ON, CUE and SELECT button and meter. Each fader also has a rotary knob that can be selected in banks to control the GAIN, PAN or assigned to a parameter. To the right of the channel section are button to choose layer for the channels, Stereo inputs, DCAs, or Custom layers.
The right side of the console contains the last channel bank that allows control of DCA layers, Stereo ins, or six Custom layers.
The far right of the surface sports the master section that includes two faders, each with ON, CUE, SELECT, meters and a rotary knob.
The master faders are also customizable able to become inputs, outputs, DCAs, etc. The front face is outfitted with a headphone jack with volume, as well as a talkback XLR input with level knob.
On the rear panel, there are eight omni XLR mic/line inputs, eight omni XLR outputs, the mini-YGDAI I/O card slots, an AES/EBU digital output, 15-pin GPI connector, two BNC connectors for word clock in/out, MIDI in/out, a pair of Ethercon connectors for the Dante network, an RJ connector for the computer network connection, and XLR lamp connectors.
A locking IEC connector is supplied for power but the CL5 also sports a multi-pin connector for the optional (PW800W) backup power supply.
If the internal power supply has a problem, the PW800 will seamlessly take over.
Overall the CL5 has a great look and feel. The fader caps are a new design from Yamaha and are very comfortable, and all the knobs are well laid out and easy to reach. The buttons, knobs and faders have a solid feel and should last for a long time.
Further, the user interface is easy to learn and get around on without spending a lot of time in the manual. The touchscreen allows multiple button selection by simply sliding your finger across the screen, and rotary knobs can be depressed to recall a screen.
Building an effects rack on the touch screen. (click to enlarge)
The console offers a few new tools, including a premium rack that features VCM analog circuit modeling technology, and some Rupert Neve Designs EQ and compressors.
For my evaluation, Yamaha also sent along a RIO3224-D stage box. This unit provides 32 inputs and 16 outputs via XLR, as well as four AES/EBU outputs in a 5RU package. The stage box offers both a Dante primary and secondary connection point that allows daisy chaining additional boxes or the ability to connect a network in a redundant fashion.
In The Field
After checking out the console in my shop and getting comfortable with the patching, I took it out on some gigs. The first was the typical corporate type show for my company, a few presenters with some playback. With all of it’s capabilities, the CL5 was overkill, but the smaller CL1 would be perfect for these smaller (but high end) types of events.
Because I was using only one hardwired microphone as a podium backup, I didn’t feel the need to use the Rio stage box and operated the console as a stand-alone desk. I patched the Omni inputs into some channels and placed the wireless rack next to me at front of house.
I also patched in the outputs from a video deck as well as a computer for walk in/out music.
Interfacing the video deck and computer was easy because I carry a bunch of adapters and DIs to every gig, but I wished that the console also offered a stereo pair of RCA and a SPIDIF input to make it easier to interface “prosumer” gear.
This is understandable because these types of connections aren’t required by everybody.
While I brought a small rack with an EQ for the mains, I just patched a digital EQ inside the console to the mix and used that.
The second gig had a corporate party band playing in a ballroom. I set up the RIO3224-D onstage and ran a single Cat-6 run of about 120 feet to the console. Not being completely familiar with the CL5, I accidentally patched the mix outputs to 2 and 3 – not outputs 1 and 2 – but the great thing about the RIO is that it features signal lights above the output XLRs.
Troubleshooting was as easy as following the flashing lights. I quickly swapped my output cables and was ready for the gig.
Because there wasn’t a lot of time for sound check I didn’t get to really set up a lot of effects, but during the gig, it was easy to go to the menu, choose a new effects unit, assign it and adjust parameters all on the fly.
Front and back views of the Rio3223-D stage box. (click to enlarge)
When word got out that I had a CL5 in the shop, lots of folks stopped by to check it out. Everyone seemed impressed and had nothing but good things to say about it. It sounds fantastic and the new premium rack was a big hit.
Overall the CL5 is an exceptional console. The compact size makes it easy to transport, and it doesn’t take up a lot of space at front of house while still offering up tons of features. It sounds great and is easy to get around on. My only dislike, in fact, is that I have to send it back.
U.S. MSRP for the CL5 is $27,499; the RIO3223-D stage box is $8,499, and the RIO1608-D is $4,799.
Hosa Technology Debuts USB-200FB Series High-Speed USB Cables With Pivoting Connector
Design facilitates space-saving right angle connections
Hosa Technology has announced the introduction of the USB-200FB Series high speed USB cables with pivoting A connector designed to conserve valuable space in tight surroundings.
Available in 3-, 6-, and 10-foot lengths (USB-203FB, USB-206FB, and USB-210FB), the new USB-200FB USB cables feature a pivoting Type A connector—the end that typically connects to a computer—that can be set to either straight or right angle positions.
This enables one to use the cable in its straight orientation when space permits and in a right angle position in cramped quarters.
In its right angle position, the computer would typically be placed on an elevated laptop stand (so the cable can hang off the side) or at the edge of the work surface.
For DJ production rigs commonly consists of a laptop computer, mixer, turntables, and a digital controller, the new USB-200FB USB cables can be a valuable means of making connections in an environment where space is frequently limited.
To ensure interoperability with a wide range of USB peripherals, the new Hosa USB-200FB USB cables are fully compliant with the USB 2.0 serial bus interface standard and are backward compatible with the USB 1.1 standard. They also support burst data transfer rates up to 480 Mbps.
As a result, the new cables are a good option for connecting an audio interface, USB microphone or instrument, or most computer peripherals to a PC.
“The new Hosa USB-200FB USB cables provide DJ’s, musicians, and others with a valuable means of making equipment connections in the cramped spaces they frequently find themselves working in,” says Jose Gonzalez, Hosa Technology product manager. “At Hosa, we’ve been providing the cables musicians and audio pros require for over two decades, and these new USB products address a common challenge we’ve all encountered at one time or another.
“All three cable lengths feature our unique connector pivot design, exhibit superior workmanship throughout, and carry pricing that is comparable to most common USB cables lacking these features. I’m certain these new USB cables will be well received by our industry.”
The new Hosa Technology USB-200FB USB Series cables are expected to become available in January 2013.
Radial Introduces The Gold Digger Passive Microphone Selector
Engineers can quickly compare and select the best-sounding microphone for the application
Radial Engineering has introduced the Gold Digger, a unique device that enables the studio engineer to quickly compare and select the best-sounding microphone to suit the character of a particular voice.
Radial president Peter Janis explains: “Capturing the essence of a voice is critical during the recording process. This is best accomplished by selecting the most appropriate microphone and suitable mic preamp. But setting up an ‘honest’ comparison between microphones can be difficult due to the time lapse involved when routing signals and discrepancies between mixer channels.
“The Gold Digger solves the problem by routing four microphones to a single output via a ‘straight wire’ signal path,” he continues. “In other words, there are no buffers or any form of gain stage in between the microphone and the output, thus assuring a color-free signal transfer without distortion or artifact.”
The Gold Digger includes four “radio-style” switches to ensure only one microphone will be activated at any one time. Phantom power (48-volt) is generated and managed inside the unit to ensure switching between mics will be quiet and pop free.
One simply plugs in the microphones, activates phantom power for condensers and then sets the trim control so that all mics produce the same output level. The “live” mic is activated by selecting the desired channel.
The Gold Digger will start shipping in November 2012. Estimated retail price: $400 (USD).
The Gold Digger recently debuted at the 133 AES convention in San Francisco and picked up a “Best of Show” award along with Radial’s new Cherry Picker preamp selector.
Lowell Introduces Rack-Mount Power Panel For Hardwired Connection
Offers eight 15-amp power outlets, built-in surge protection and six-foot non-metallic flexible conduit
Lowell Manufacturing Company has introduced the ACR-1508-S-HW, a new rack-mount panel that has eight 15A power outlets in the rear as well as a six-foot non-metallic flexible conduit for secure connection to the power source.
The flexible conduit can be trimmed to the exact length needed and is much easier to route through tight spaces than a traditional metal-clad whip.
The new hardwired panel measures 19 x 9 x 1.75 inches (w x d x h) and includes built-in surge protection.
Two LED status indicators in front allow users to quickly confirm that power is flowing and surge protection is active.
Extron Introduces New Six Input HDCP-Compliant Scaling Presentation Switcher
Enhanced audio features include six stereo inputs, fixed and variable stereo outputs, two mic/line inputs with ducking and 48 V phantom power
Extron Electronics has introduced the IN1606, a six input, HDCP-compliant video scaler that includes four HDMI inputs, two universal analog video inputs, and two simultaneous HDMI outputs.
The IN1606 accepts a wide variety of video formats including HDMI, HDTV, RGB, and standard definition video.
It features an advanced video scaling engine with 1080i deinterlacing and Deep Color processing to deliver uncompromised picture quality for output resolutions up to 1920x1200 and 2K.
Enhanced audio features include six stereo inputs, fixed and variable stereo outputs, two mic/line inputs with ducking and 48 V phantom power, plus HDMI audio embedding and de-embedding.
Designed for professional AV integration, the IN1606 offers a complete AV switching system with flexible control options including Ethernet, RS-232, and USB.
“The IN1606 combines multi-input switching and signal format flexibility with advanced scaling functionality to deliver a high performance, one-box solution for today’s presentation systems,” says Casey Hall, vice president of sales and ,arketing for Extron. “Powerful audio capabilities are also valuable features of this scaler, enabling audio source integration and fine-tuning to suit the needs of each installation.”
To enhance and simplify integration, the IN1606 features SpeedSwitch Technology, which provides exceptional switching speed for HDCP-encrypted content.
EDID Minder and Key Minder automatically manage EDID communication and HDCP key negotiation between input and output devices to ensure reliable operation.
The IN1606 also provides immediate visual confirmation and real-time HDCP status verification, offering valuable feedback to system operators and helpdesk support staff.
With HDMI audio embedding and de-embedding, the IN1606 can insert analog input audio signals onto the HDMI output or extract embedded HDMI audio signals. Audio breakaway allows the analog audio channels to be separated from corresponding video signals so that the audio channels can operate as an independent switcher.
The IN1606 also provides complete control of advanced audio configuration settings, such as audio gain, attenuation, mixing, and ducking through an intuitive Graphical User Interface.
AES & OCA Alliance To Collaborate On Open Control Architecture Networking Standard
Goal is to produce a public, open, and royalty-free communications protocol standard
The Audio Engineering Society (AES) and the OCA Alliance have jointly announced that an AES standards project has been founded to consider OCA, the Open Control Architecture, for a public standard for control and monitoring of professional media networks.
The goal of this project, identified as AES-X210, “Open Control Architecture (OCA),” is to produce a public, open, and royalty-free communications protocol standard for reliable and secure control and monitoring of interconnected audio devices in networks of 2 to 10,000 elements.
When the standard is complete, it is hoped that OCA will find broad acceptance in the media systems industry, and open a new era in standardized, interoperable control of devices from diverse manufacturers.
OCA will be a control and monitoring standard, not a media transport standard. It is intended to operate seamlessly with a wide range of media transport standards such as AES-X192, “High-performance streaming audio-over-IP interoperability” - currently in development - and IEEE AVB.
Together, these standards will offer a path to complete network application solutions for future media networks that include both media transport and system control functionality.
OCA is substantially based on work done in the early 1990’s by the AES24 project, a pioneering effort in network system control. Although AES24 never reached full standards status, it offered a number of advanced concepts, which have found their way into various developments over the years.
In the coming months, the AES-X210 task group, part of AES working group SC-02-12 on Audio Applications of Networks, will be meeting to render the current OCA 1.1 specification into standards form and to shepherd its processing through the AES open standards process. Interested individuals are encouraged to participate.
The full standards participation policy is available from the AES Standards website. (Direct link is here.) During this period, the OCA Alliance will undertake various initiatives to support the work of AES-X210, and to define proposals for future extensions to the standard.
AES Standards Committee Chair Bruce C. Olson remarks, “AES Standards has been deeply involved with standards for digital audio since starting work in 1977. The AES-X210 project to standardize OCA takes us to the next important milestone by integrating control with transport of audio over a variety of networks.
“Combined with a number of other projects in this exciting area of audio the AES continues to lead the world in audio standards, which enable a thriving marketplace for compatible products from competing manufacturers. We are very pleased to collaborate with the leading manufacturers in the OCA Alliance on this standards project.”
Extron Introduces New Economical Wall Mount HDMI Twisted Pair Extender
Extron Electronics is pleased to introduce the DTP HDMI 230 D, a Decora-style transmitter and receiver set for transmission of HDMI, audio, and bidirectional RS-232 and IR control signals up to 230 feet (70 meters) over a single CATx cable.
Extron Electronics is pleased to introduce the DTP HDMI 230 D, a Decora-style transmitter and receiver set for transmission of HDMI, audio, and bidirectional RS-232 and IR control signals up to 230 feet (70 meters) over a single CATx cable.
The HDCP-compliant extender provides an economical and effective means for extending HDMI with embedded multi-channel audio from HDMI-equipped devices.
In addition, the DTP HDMI 230 D accepts analog stereo audio signals and digitizes them for simultaneous transmission over the same twisted pair cable.
The wall-mount design and remote powering of either unit make the DTP HDMI 230 D ideal for extending HDMI, audio, and bidirectional control while offering aesthetically pleasing integration in space-challenged environments.
“The DTP HDMI 230 D increases system flexibility with a convenient wall-mount design that enables AV system designers and integrators to add inputs and outputs exactly where they are needed,” says Casey Hall, Vice President of Sales and Marketing for Extron. “This extender also streamlines integration by sending HDMI video, audio, and bidirectional control signals over a single twisted pair cable.”
This extender simplifies the incorporation of analog and digital audio signals. It allows a direct analog audio connection from devices with stereo output, such as desktop computers or laptops, and provides balanced and unbalanced audio output from the receiver.
In addition, the DTP HDMI 230 D continuously maintains DDC communication of EDID and HDCP between a source and display for reliable operation, ensuring direct compatibility and optimal signal transmission between devices.
The DTP HDMI 230 D is compatible with CAT 5e, CAT 6, and CAT 7 twisted pair cable, and can be used as a point-to-point solution or integrated with an HDMI matrix switcher to extend inputs or outputs to remote locations. It supports signal resolutions up to 1080p/60 or 1920x1200 and supports HDMI specification features including data rates up to 6.75 Gbps, Deep Color up to 12-bit, 3D, HD lossless audio formats, and CEC.
For added flexibility, either the transmitter or receiver can be powered over the original twisted pair cable, allowing both devices to share one external power supply. Both devices are available separately, and may be mixed and matched with the desktop DTP HDMI 230 transmitter or receiver to suit the installation requirements of a specific application.
Audinate Joins Open Control Architecture Alliance (OCA Alliance)
Alliance goal is to create an open public control standard for professional media network system
Audinate, inventor of Dante networking technology, has joined the OCA Alliance as an associate member.
The OCA Alliance was formed by professional audio companies who work in different product markets and represent a diverse cross section of vertical market positions and application use-cases.
Founding OCA Alliance members Bosch Communications Systems, Salzbrenner Stagetec Mediagroup, TC Group and Yamaha Commercial Audio are Audinate partners and have incorporated Dante as their digital media networking solution.
The OCA Alliance goal is to create an open public control standard for professional media network systems and to ensure the standardization of the Open Control Architecture (OCA) for a wide interoperability of professional audio equipment.
The alliance was formed to complete the technical definition of OCA, then to transfer its development to an accredited public standards organization.
Audinate Dante provides high performance digital media networking over both 100Mbps and 1Gigbit Ethernet. Dante provides low latency, sample-accurate playback synchronization, high channel counts and is plug and play. It has been licensed by over 60 OEMs customers across the AV industry.
“We have been following the developments of the OCA Alliance since its inception and feel that being an associate member is for us a good opportunity to be involved in this important alliance,” says David Myers, Audinate COO. “We are pleased that this initiative to have a common control architecture for AV manufacturers has gained so much momentum. “
“The OCA Alliance is very happy to welcome Audinate as an associate member. As a provider of an innovative and widely adopted media networking architecture, Audinate’s participation in The Alliance will be of great benefit for the standard and its adoptees alike,” says Bill Scott, OCA Alliance chairman. “Audinate has been very supportive of OCA, and we are happy to have taken this support to the next level by adding Audinate as a member.”
Updates flagship multi-track recorder with the ultimate sound quality, higher speed, extreme reliability, and modern flexibility
iZ Technology Corporation, manufacturer of innovative music and audio products, has launched RADAR 6 at the 133rd AES Convention in San Francisco. RADAR, the premier choice for hard disk multi-track recording and playback employed by recording studios, scoring stages, theaters, and post production houses around the world, has been updated with new features and improved speed, storage technology, editing, audio performance and user interface.
RADAR has long been the recording platform that offers the highest sound quality on the market. Built with world-class converters, Adrenaline DR technology, dual digital/analog power supplies, near-zero jitter, and near-zero latency up to 192kHz, RADAR systems transcend the digital barriers of sonic quality.
Thanks to RADAR 6’s new storage architecture, recording and moving files is fast and easy. You can record 24 tracks at 192 kHz to a 64 or 128 GB SD card and plug it directly into your laptop for use with your favorite DAW. Record and/or copy and deliver tracks in seconds on a USB 3.0 thumb drive. Or record directly to RADAR’s high-speed solid-state drives for maximum performance. RADAR’s open storage architecture gives you the most options.
New features of RADAR 6:
Sound Quality: World-class Classic 96 and Ultra Nyquist converters and the new Adrenaline DR technology
Speed: Direct SATA recording for over 2 times faster cueing editing and data transfers
Storage: Solid state SATA, USB 3.0, thumb drives and remote network access make recording and file transfer fast and flexible
Portability: nearly 40% shorter and 14 lbs lighter than RADAR V
Improved user interface: full front panel controls with tactile transport keys and a comprehensive high resolution multi-page touch screen supporting industry standard wide screen resolutions
Editing: Comprehensive editing from the front panel touch screen as well as a virtual meter bridge, touch-screen macro keys, multiple mark in/out markers for batch exports and audio CD creation from a single project
Acoustic Noise: control room ready with super quiet low speed power supply fan and all solid-state storage drives
With choices of 3 different converter card designs available in 8, 16, or 24 channel configurations, 4 multi-channel digital I/Os, and a wide variety of backup and recording drive options, RADAR is the most configurable multi-track recorder available today.
Pricing and Availability: Starting at under $8,600 USD.
Zeehi CueCast Provides User File Conversion For DiGiCo SD7 Digital Console
Raises number of supported console models to five
Zeehi announces that the CueCast digital mixing console user file conversion service will support the DiGiCo SD7 platform beginning mid-November 2012.
This raises the number of supported console models to five in CueCast’s latest Beta release.
CueCast enables sound engineers, hire companies and production managers to quickly and easily convert complex show files between different digital audio mixing consoles.
“We’ve had a very positive response from the pro audio community since we released the first CueCast Beta version in July,” states Danny Abelson, co-founder of Zeehi. “A growing number of engineers have been converting their files at our site, and new users from around the world are signing up for the service every week.”
CueCast gives every new user two free fully-featured conversions, allowing anyone who’s interested can evaluate the service with no cost or commitment.
“We’ve been working closely with several of the large hire companies,” Abelson adds. “One of their most frequent requests was that we add support for the DiGiCo SD7, since they have so many of these desks in their inventories. We were happy to oblige, and our development team is doing a great job implementing the new console. We’ve also made significant enhancements to the user interface.”
In its latest version, CueCast converts user files between Avid, Yamaha PM5D, and now three DiGiCo models: SD7, SD8 and SD10. The company plans to develop solutions for every major console model.
The web-based CueCast service solves a fundamental problem in audio engineering; transferring complex user settings from one console to another without the time-consuming headache of entering those settings manually.
“Converting show files on CueCast is reliable and takes just three easy steps,” Abelson says. “Simply upload your file to the secure Cuecast site, then specify the format you need, and download the converted file for installation in the new console. CueCast stores your files on a secure server for safe-keeping and future use.”
He continues, “We are helping people improve the way they work by making the process of moving user settings to a new console fast and easy. We can bring the biggest benefit to the engineering community by expanding the number of different console models we support. Adding the SD7 was a significant step. Soon we’ll be announcing support for more Yamaha models, including the new CL range, as well as the Soundcraft Vista and Midas desks.”
The current Beta release of CueCast converts the most commonly used console features and functions including busing, sub-group assigns, control group assigns, routing, labeling, mutes and mute groups, EQ and dynamic in/out settings, aux. send on/off/and assigns, and effects and matrix on/off and assigns. Future releases will support variable settings, snapshots and many other features.
How To Find Problems That Degrade Performance In Balanced Audio Equipment Interfaces
The problems fall into two broad categories
Previously I have explored the theoretical aspects of balanced interfaces and de-bunked the widespread myth that equal and opposite signal swings are somehow responsible for its noise-rejection properties.
In this column, we will explain how to find most problems that degrade performance in real-world systems.
The problems fall into two broad categories.
The first is design errors in the equipment itself, most often the “Pin 1 problem” that couples shield current noise into the audio.
The second is problems caused by the type and/or routing of the audio cables themselves.
Since cable shields are tied to Pin 1 in XLR connections, this equipment design error was dubbed the “Pin 1 problem” by Neil Muncy in his famous 1995 AES Journal paper. This form of common-impedance coupling has been inadvertently designed into a surprising number of well-known products with balanced interfaces.
As Neil says, “Balancing is thus acquiring a tarnished reputation, which it does not deserve. This is indeed a curious situation. Balanced line-level interconnections are supposed to ensure noisefree system performance, but often they do not.” (1)
The Pin 1 problem makes the cable shield contact act as a very low-impedance signal input! Shield current, consisting mainly of power-line noise, is allowed to flow in internal wiring or circuit board traces shared by amplifier circuitry. Because these wires or traces have electrical resistance, tiny voltage drops are created and amplified to appear at the equipment output.
When this problem exists in systems, it can interact with other noise coupling mechanisms to make noise problems seem illogical and confounding. This problem can afflict video and other unbalanced interfaces, too.
Fortunately, a simple test will reveal the Pin 1 problem. The “hummer” (Figure 1) is based on an idea suggested by John Windt (2). This simple device, whose schematic is shown here, forces an AC current of about 50 mA to flow through the potentially troublesome shield connections in the equipment under test.
Figure 1: The “hummer”
In property designed gear, this causes no additional noise at the equipment output. The 12-volt transformer should be rated for about 50 mA. The optional LED (and 1N4001 diode) are used simply to confirm that current is indeed flowing.
Using the hummer, step by step:
1. Except for the output being monitored, disconnect all other cables and any other chassis connections, such as rack mounting, from the equipment under test.
2. Power up the equipment.
3. Meter (and listen to, if possible) its output. The only noise should be white noise or “hiss.” Try various settings of operator controls to familiarize yourself with the noise characteristics of the device under test without the hummer connected.
4. Connect one hummer lead to the device chassis and touch the other lead to the shield contact of each input or output connector. If the device is properly designed, there will be no output hum or change in the noise floor.
5. Test other potentially troublesome paths, such as from an input shield contact to an output shield contact or from the safety ground Pin of the power cord to the chassis or shield contacts. No such path should result in noise.
Rarely, shield contacts are not tied directly to chassis or safety ground and the hummer’s LED will not glow.
This is cause for concern, especially on an output—output shields must always be grounded (3).
In any case, try to find out where the manufacturer has actually tied the shield connection.
Easily constructed test adapters or “dummies” allow the system to test itself and pinpoint the exact entry point of noise or interference.
By temporarily placing the dummies at strategic locations in the interface, precise information about the nature of the problem is also revealed.
Figure 2: Correct wiring for test adaptors (dummies).
The tests can specifically identify:
• Shield-current-induced coupling in cables
• Magnetic or electrostatic pickup by cables of nearby fields, or
The dummies are made from standard connectors wired as shown in Figure 2—do not connect the metal shell to Pin 1. Remember that they do not pass signal.
Each signal interface is tested using the following four-step procedure:
Unplug the existing cable from the input of Box B and plug in only the dummy.
Output quiet? No—The problem is either in Box B or further downstream. Yes—Go to next step.
Leaving the dummy in place at the input of Box B, plug the existing cable into the dummy.
No—Box B has an internal “Pin 1 problem.” The hummer test can confirm this. Yes—Go to next step.
Remove the dummy and plug the existing cable into the input of Box B. Unplug the other end of the cable from Box A and plug it into the dummy. Be sure the dummy doesn’t touch anything conductive.
No—Noise is being induced in the cable by an external magnetic or electric field. Check cable shield connections at both ends and/or re-route the cable to avoid interfering fields. Yes—Go to next step.
Leaving the dummy in place on the existing cable, plug the dummy into the output of Box A.
No—The problem is shield-current-induced noise or SCIN. Replace the cable with a different type or take steps to reduce current flow in the shield (4). Yes—The noise must be coming from the output of Box A. Perform this 4-step test sequence at the next upstream interface.
Bill Whitlock has served as president of Jensen Transformers for more than 20 years and is recognized as one of the foremost technical writers in professional audio.
REFERNECES (1) Muncy, N., Noise Susceptibility in Analog and Digital Signal Processing Systems, Journal of the Audio Engineering Society, June 1995, pp. 435-453.
(2) Windt, J., An Easily Implemented Procedure for Identifying Potential Electromagnetic Compatibility Problems in New Equipment and Existing Systems: The Hummer Test, Journal of the Audio Engineering Society, June 1995, pp. 484-487.
(3) Whitlock, B., Balanced Lines in Audio Systems: Fact, Fiction, and Transformers, Journal of the Audio Engineering Society, June 1995, pp. 460-462.
(4) Brown, ]. and Whitlock, B., Common-Mode to Differential-Mode Conversion in Shielded Twisted-Pair Cables (Shield Current-Induced Noise), Preprint 5747, Audio Engineering Society 114th Convention, March 2003, Amsterdam
Antelope Audio has introduced the Orion 32, the world’s first 32-channel AD/DA converter and audio master clock in a 1U rack.
The new device supports both MADI and USB interfaces, clocked by Antelope’s 64-bit Acoustically Focused Clocking (AFC) technology.
Orion 32 allows 192 kHz I/O streaming of 32-channel digital audio through its custom-built USB chip, which provides simple connectivity to any USB-enabled DAW or computer.
The converter also provides 32 channels of 96 kHz audio through its fiber optic MADI I/O connections, which can be used to connect with any suitably equipped MADI device.
Orion 32 also supports ADAT protocol by offering 16 I/O channels, for even greater compatibility with a large number of audio devices. The multi-channel converter inputs and outputs pass the analog signal through eight D-SUB 25 I/O connectors.
In addition to being an extremely high quality audio converter, Orion 32 is also an audio master clock.
The Orion 32 employs Antelope’s proprietary 4th generation of AFC and oven controlled oscillator for accuracy and reliability.
The four word clock outs, together with the 10 MHz input, make Orion 32 suited to be in the center of any project or high-end studio.
“The multi-channel Orion 32 is the logical step, following our highly acclaimed stereo AD/DA converter Eclipse 384 we launched last year,” says Igor Levin, Antelope Audio CEO. “We have developed this device in response to customers who have demanded Antelope’s signature-quality sound, but in a multi-channel device.
“Orion 32 is ideally suited to a sound engineer’s needs for high quality, clean and transparent conversion, combining the efficiency of MADI and USB interfaces with the perfection of Antelope clocking.”
The routing feature allows sound engineers to, for example, use the AD conversion and output the signal simultaneously both through the MADI and USB interfaces.
The device is managed through a user friendly desktop application available for both Windows and OS X. Moreover it is equipped with five preset buttons for fast and easy recall of favorite settings.
Orion 32 is packaged in a 1U enclosure with an eco-friendly power consumption of 15 watts.
• Antelope Audio precise AD/DA conversion technology
• Up to 192 kHz on 32 channels I/O via custom-built USB chip
• 32 channels I/O via Fiber Optic MADI I/O connections
• Eight D-SUB 25 connectors for AD and DA
• 16 channels on 4 ADAT input and output connectors
• Antelope’s renowned 64-bit Acoustically Focused Clocking with Atomic input
• Antelope’s proprietary Oven Controlled Oscillator for supreme clocking stability
• Four word clock outputs and one word clock input
• Five presets for fast and easy recall of favorite setups
• 1U rack size with power consumption of 15 Watts and very low heating levels
• A user friendly desktop application available for both Windows and OS X
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