Transcript
November 2014
APPLICATION FOCUS Distance Conferencing Boardroom based distance conferencing usually consists of microphones, speakers, amplifiers, echo cancellers and a control interface, all connected to a codec of some sort. This month we will take a look at some HARMAN products that could be used in such applications.
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Firstly, what needs require addressing in a distance conference application? Pick up of the talkers voice in the conference room, we will call this the near side. Relay of the other distant parties voice, we'll call this the far side, to be heard in the near side conference room on the loudspeakers. We need to employ a device called an echo canceller to, as the name may suggest, remove echo detected in the microphone pick up path before it gets transmitted back to the far side. If we did not use this, then the far side audio, as it was heard through the speakers, would be picked up by the mics on the near side conference room table and sent back to the far side, causing the echo. If you hear echo in a conference room, then it will be because there is no echo canceller at work, or the system has been deployed poorly.
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So, we need microphones to pick up the near side talkers, a DSP to provide the Acoustic Echo Cancellation (AEC), an amplifier, and speakers so we can hear the far side (and possibly the near side too if that is required). Third party Codec's can be used to transmit and receive the audio, or a regular analogue phone line could be used.
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We can address all this with the following items; AKG MB-3 discreet boundary microphones fitted into the conference room table; a BSS Soundweb London AEC DSP, there's the echo canceller, automixer, and speaker processor; a Crown ComTech CT8150, an 8 channel energy efficient, convection cooled amplifier; Distance Conferencing
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JBL Control 47LP ceiling speakers, a premium model, and paintable to match the room colour if required. That's the basic components.
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Lets say the client has some additional requirements. You can't lay analogue cable across the conference room floor, or dig up his floor for that matter either.
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We could use a BSS BLU-BIB to transport the microphone audio from the table to a BLU-101 in the rack room over CAT5e. Usually boardroom tables are equipped with multiple CAT5e sockets all running back to a patch panel in the building. BSS BLU-Link transport can run up to 100m between devices, so this is pretty much always enough, and you can extend with Fibre media convertors if it comes to it. If not, then a single piece of CAT5e is always going to be easier to install than pulling an analogue mulitcore in!
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Your client may also want to replay local AV, and share this with the far side too. We can employ a pair of JBL CBT-50LA full range column speakers either side of the projection screen or flat panel monitor. These speakers produce a controlled vertical beam and are excellent for a throw of around 10m or so, and produce a very uniform level across a wide frequency range in this scenario.
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A typical schematic outline of how this would go together is shown below:
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Lets take a closer look at each component in the signal path in this application.
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The AKG MB-3 is a very low profile, through the table mounting boundary microphone, as you can see on the right, measuring less than 2.5cm in diameter on the table when fitted.
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It is a supercardoid design, with great rejection to the rear of the mic, around 20dB down, making this ideal for use on a conference table where there are talkers on both sides of the table. Best capture is within a 90 degree plane from the table surface. It exhibits a flat response with a presence lift of 3dB around 5kHz to assist in intelligibility.
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To save digging up the clients precious boardroom floor to lay analogue multicore, we will connect these microphones to a BSS Soundweb London BLU-BIB (Break In Box), shown below. The BLU-BIB features up to 48dB of mic preamp gain for each of it's 8 channels, and individually switchable 48v phantom power. It's a half rack sized device and ships with brackets to facilitate under table mounting. Pre-amp gain and phantom power is set via the front panel, and then locked out. Mics are connected at the rear and a single, or pair of CAT5e cables are plugged into the rear BLU-link ports. Use a single cable if you don't require redundancy protection for a failed cable, use 2 if you do!
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In this example we are using only 8 microphones around the conference table, but if we wanted to use more, we can add BLU-BIB's for every additional 8 microphones. Of course, should the client want to connect a laptop for audio input to the system, BIB inputs could be used for that audio too.
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From here we can follow our signal path to the BSS Soundweb London BLU-101 AEC equipped DSP. When using AEC it is best practice to use one AEC processing channel for each microphone present in the conference room. The BLU-101 has twelve AEC processing channels, and so is fine for our purposes here, and indeed has some AEC processing to spare, useful if the client decided to add a wireless microphone system for presenting at a later date.
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The BLU-101, like all other Soundweb London DSP devices has a completely open internal architecture, where the programmer can design the audio signal path in any way that is desired. This makes it a very flexible platform, and in this instance will allow us to employ useful processing objects such as high pass filters, microphone EQ, gated automixer and a mix minus matrix for the local reinforcement. All in addition to the Acoustic Echo Cancellation which the BLU-101 also provides.
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An example of this layout internally in a BLU-101 is shown below, as designed using HARMAN's Audio Architect software platform, though if you are new to programming DSP devices, you can download this example file from this location to help you get started:
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https://dl.dropboxusercontent.com/u/40174272/Distance%20Conferencing.audioarchitect
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Lets take a look though the signal path for the local, or near side, audio firstly.
The near side audio arrives via the analogue inputs, from the PC in the rack and the input from the far side that arrives via the codec. The microphone audio arrives via BLU-Link from the BLU-BIB fitted under the conference room table. The signals are split via wire tags and then pass through a high pass filter to remove unwanted low frequency information from the microphone pick up path, and a parametric EQ which we can use to colour the microphone and PC and Codec audio if required, and a gated automixer is next in the path of the microphones.
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The gated auto mixer, as the name suggests, gates, or unmutes, microphone channels as it detects audio present on those channels. There are various settings to adjust precisely when this occurs, including activating voice band filters to prevent the mixer from opening when people at the table click pens, shuffle papers or move computers around for example. We can also set a maximum number of microphones the mixer will allow to be open at the same time, and a priority associated with each microphone, plus assorted other options. One of which is to activate the direct outputs from the gated automixer instead of using the summed mix of all active outputs. Using the direct outputs allows us to make use of the gating ability of the mixer, and still retain each microphone signal on it's own for processing in the next object in our signal chainthe matrix mixer.
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The client may have desired to have the local microphones reinforced via the ceiling speakers. This is ok, but this can have the potential to generate acoustic feedback in the near side room- do not confuse this with Acoustic Echo Cancellation, they are not the same thing! In order to minimise feedback, but to hear these mics in the room, we use the matrix mixer to remove the mic's audio from the speaker nearest it - ie mic 1 is not routed to speaker circuit 1, and so on.
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In our example above, mic 1 is active and highlighted in green and audible on the speaker circuits shown in green, the nearest speakers to microphone #1 receive no audio from this microphone and are shown in red.
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Usually mix minus arrangements such as these are used in larger conference rooms, where it would be difficult for those in attendance locally to hear the talker without reinforcement from the loudspeaker system. In smaller rooms it's generally not required. We also use the matrix mixer to add the local PC audio and received far side Codec audio and direct these to our chosen speakers.
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At this point we also split the PC and Codec audio and connect these via a simple summer to our AEC reference object. AEC is a complicated algorithm that essentially listens to each channel's audio input, and compares that to the audio connected to the reference object for the same AEC channel. Any audio which matches is removed from the audio path as it passes through the AEC object. In our application this means that audio from the PC and far side audio from the Codec get removed from the microphone pick up path, leaving just the speech. That is just what we want. There are all sorts of other applications for AEC, but we will leave that for another time.
A detailed video describing what AEC is and how it works is available from the BSS website at this address:
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http://www.bssaudio.com/en-US/videos/play/t6pW3r8ouEY
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The audio to be transmitted to the far side consists of the table mics, and the local PC. This allows media sharing with the far side if a computer (or other) audio source needs to be shared between both locations. The signal path is similar to that for the local reinforcement, minus the codec far side audio of course. This time a simple mix from the gated auto mixer is sufficient to send to the far side, via the analogue outputs and a codec.
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The audio now splits, the far side audio mix heading to the codec for transmission to the other side, and the local audio is supplied to a Crown Comtech CT8150 amplifier.
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The CT8150 is ideal in this application as it's small, just 1RU and silent in operation because it has no fans. The convection cooled design is possible due to the use of a proprietary Crown Drivecore IC amplifier, developed by HARMAN and featuring 5 patents. The highly efficient design means the amplifier doesn't get warm enough to need a fan, and is smart enough to switch unused or idle channels into standby mode when no audio is detected on that channel for more than 30 mins. Power usage in standby is less than 1w. Input routing is flexible too via a set of rear panel DIP switches to set input to output routing.
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This energy efficient amplifier supplies 125w per channel into any 4-8ohm load. An optional low cost low saturation step up transformer pack is available if we need to drive a 100v/70v speaker line. In this application we will be using low impedance speakers from JBL- some ceiling mounted Control 47LP models and a pair of surface mounted CBT50-LA models mounted either side of the display screen in the room.
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Lets look at the CBT50-LA firstly. The 500mm tall column is available in black or white finishes, and indeed can be painted to match the room decor. It's physical size makes it a natural fit alongside a LCD or Plasma display.
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CBT is an abbreviation for Constant Beamwidth Technology, an analogue beam forming and amplitude shading method employed by JBL and patented to accomplish consistent vertical coverage without the narrow vertical beaming and out-of-coverage lobing that are typical of straight form-factor passive column speakers.
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CBT speakers, such as the CBT50-LA in this application can be modelled in a simple freely available software program to see if they are suitable for the space. The software is JBL CBT Calculator, and you can take a look at a video overview of it here:
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http://www.jblpro.com/www/products/installed-sound/cbt-series#.VGLR_PmUd8E
Modeling the pair of CBT-50LA in our conference room with a 8m long table turns out like this:
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Yo u c a n c l e a r l y s e e t h e converging SPL at higher frequencies, no matter the listeners position at the table. Each line trace represents a different position on the 8m boardroom table. If we examine the SPL variances between 80Hz and 1250kHz, and those between 2500kHz and 4000kHz, we can see that very little variation occurs, the largest being +/-0.8dB.
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The result of this being that the listeners closest to the speaker do not get deafened and the listeners furthest away do not struggle to hear. Everyone hears the audio at a very similar level.
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In our application we will be using these CBT50-LA speakers to provide media replay from a PC or other local, or remote program audio source. Usually this will be in conjunction with video, the location of the CBT50-LA's either side of the screen serving to provide a directional image for the sound, instead of the sound from a video source being replayed via the ceiling speaker grid, above the listener.
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Which leads us on nicely to the ceiling speakers. We have chosen the Control 47LP, from JBL's premium Control 40 series. JBL manufacture a huge amount of ceiling speakers, and choosing which one is appropriate for any particular application can be challenging. In any application it is important to correctly asses the clients needs and expectations. A poor quality choice in speakers may come back to haunt you later!
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Helpfully, JBL have provided some tools to assist you in selecting a suitable loudspeaker. One of these tools is the simple Distributed Speaker Designer, or DSD. You can download a copy of that program here:
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http://www.jblpro.com/ProductAttachments/SetupDSD_V3.5.0.zip
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Using this software the user inputs the room dimensions and chooses a coverage variance that will be acceptable. This coverage requirement varies with the application and ranges between a 'Maximum Overlap' setting which applies an average of 1.4dB across the space, to a '2x Edge to Edge' setting where the level varies by around 10.5dB. In our instance, our client likely wants a very consistent coverage with little variance in the audio level no matter where in the room the listener is seated. A much wider variance may be acceptable in a school paging system, for instance.
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In order to find a suitable speaker, we input these values and choose 'Maximum Overlap', for a variance of 1.4dB across the 12m long and 7m wide room. We are even supplied with speaker positioning plots and spacing information.
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Choosing the speaker is simply a matter of selecting a model and observing the result showing various details, as in our example shown on the right.
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Using the Control 47LP, we can see that 8 speakers is sufficient to cover this space evenly.
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If we select a different, more cost effective speaker, say the JBL 8128, then it will be shown that many more speakers would be required to do the same job, mainly due to the dispersion characteristics of the 8128. In this instance we'd need 28 JBL 8128's instead of just 8 of the Control 47LP. Once costs of installation are taken into consideration, the initial expense of the premium sound Control 47LP may be quickly overtaken by the cost of installation time, effort and cable required to fit the more cost effective model.
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The Control 47LP loudspeakers offer outstanding pattern control for extremely consistent coverage combined with superior sonic performance. The Control 47LP features innovations such as our proprietary conical Radiation Boundary Integrator® (RBI™). Control 40 series models 47C/T, 47LP, 45C/T & 47HC incorporate JBL's proprietary conical RBI which is adapted from the groundbreaking VERTEC and VTX Series of line array toursound loudspeakers. This unique JBL patent-pending innovation combines a large diameter highfrequency waveguide with low-frequency projection apertures that work in tandem to provide a seamless integration of coverage between the two coaxially-mounted drivers. The result is extremely consistent pattern control and coverage.
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In other words, you don't need to be directly on axis under the speaker to receive full frequency coverage across the 120 degree dispersion pattern, unlike some of our competitors.
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The Control 47LP also offers us a shallow design without loss of low frequency response- the back can is just 142mm deep, ideal for fitting into ceiling voids that may incorporate many additional services.
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Control of this system can be accomplished by using Apple iOS devices running the HARMAN 'Motion Control' app, with custom user designed control panels or touch screen PC's running similarly designed panels, or depending on the actual use requirements, BSS Contrio wall control panels, or even a BLU-8v2 wall panel.
Multiple controls can be deployed to these panels. An example of a system in use at a HARMAN facility is shown below:
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The panel above clearly demonstrates the control abilities of Audio Architect when deployed on an iPad, suitable for use in a small boardroom conferencing application. The BSS Soundweb London DSP provides the control interface to a flat panel monitor and a lighting control system, for simple control via the LAN and/or a direct serial connection.
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Often these applications will require a more advanced control system to be engaged with the audio system, and likely some type of adaptive integration with video displays also. To that end we would encourage you to review our AMX solutions including a comprehensive overview of video conferencing technology here:
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http://www.amx.com/automate/learn/learntracks/#!/videoConferencing
HARMAN Audio Architect software is the subject of an Asia Pacific training program rollout. This program discusses, designs and deploys various real world scenarios and some of the techniques described in this application focus. We hope to see you at one in the near future.
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Details on upcoming training sessions in your territory and course registration is available via HARMAN's Architectural Media Systems webpages:
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http://archimedia.harman.com/en-US/training_overview
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