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Polycom EF2241Polycom EF2241 Automatic Mic / Matrix Mixer with Phone Hybrid
Expand the audio capabilities of your installed conferencing facility

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Brand: POLYCOM
Part Number: EF2241


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Related manuals
Polycom EF2241 Supplementary Guide
Polycom EF2241 Reference Guide
Polycom EF2241 Frequently Asked Question
Polycom EF2241 Software Guide
Polycom EF2241 Product Brochure

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Remote Control

RS-232 port for remote control Reconfigurable parallel logic input/output EF bus for linking multiple Vortex devices Digital bus with 5 audio busses, 48 kHz sampling rate Up to 8 Vortex devices can be linked, each device providing 4 audio signals on the bus and each EF2241 providing 1 phone audio signal on the bus

PRE-INSTALLATION

Whats Included
The Vortex EF2241 product package includes the following items: Vortex EF2241 Reference Manual Vortex EF2241 External power supply Cat 5 cable for EF Bus Telephone cable Rack mount screws (4) Phoenix connectors (16) Cable clamp and screw CDROM containing other manuals and Conference Composer software Product Registration Card

Rackmount Screws (4)

Vortex EF2241

Phoenix Connectors (16)

Cable Clamp and screw

1-Cat 5 cable for EF Bus

CDROM with control software and manuals
External Power Supply 1-Telephone cable Vortex EF2241 Reference Manual
Figure 1. Whats Included with your Vortex EF2241.

Whats Not Included

The following equipment is not included with the EF2241 product package, but may be necessary to create a completely functional system: Microphones Loudspeakers Audio cables Videoconferencing codec or other four-wire interface (optional) RS-232 remote control device (optional)
Tools Needed for Installation
Screwdriver to mount the EF2241 in your rack. Phoenix connector screwdriver
EF2241 FRONT AND REAR PANELS
Figure 2. EF2241 Front and Rear Panels 1. 2. 3. 4. 5. 6. 7. 8. LCD DISPLAY. Displays menu instructions for configuration and operation of the EF2241. DOWN BUTTON. Scrolls backward through menu items at a particular level or decreases the value of a parameter. UP BUTTON. Scrolls forward through menu items at a particular level or increases the value of a parameter. ENTER. Enters the menu and allows you to select and change parameter values. ESC. Returns to the next highest level of menus. HOME. Returns to the top of the menu structure. LEVEL INDICATOR. Indicates the level of the selected channel or parameter. PHONE CONNECT. Takes the phone line on or off hook. If you have an analog handset connected to the PHONE jack on the back panel, pushing this button will disable the PHONE jack while enabling the LINE jack (see Item 17). CHANNEL ACTIVITY LEDS. Indicates gating activity of the 4 mic/line channel inputs.

CONNECTING THE EF2241 TO OTHER EQUIPMENT

Grounding

The EF2241 has 4 mic/line inputs plus 4 line level inputs and 8 line level outputs. Each input/output is Pin 1 compatible this means that the ground pin of each input/output is tied to chassis ground. Chassis ground is connected to the input power ground.
INSTALLATION Typical EF2241 Connections
The EF2241 will typically be connected to other equipment in a single room setup as shown below in Figure 5 and Figure 6.
In 1 In 2 In 3 In 4 In A In B In C In D
Out 1 Out 2 Out 3 Out 4 Out A Out B Out C Out D

Polycom Video CODEC

Line In Line Out
Spk Out Phone Line EFBus In EFBus Out PSTN
Figure 5. Block diagram of typical EF2241 connections: a single room using one EF2241.
Polycom Video Codec, 4-wire Connection POLYCOM CODEC
Figure 6. Typical EF2241 connections. The following steps are typically used to set up the EF2241: Connect a microphone to each of the 4 mic/line level inputs. The mic/line input accepts mini-Phoenix connectors. See Connector Pinouts on page 51 for pinouts of Phoenix connectors.
Connect speaker output to loudspeaker. If RS-232 remote control is desired, connect the RS-232 REMOTE CONTROL port of the EF2241 to the remote control device, such as an RS-232 interface to a touch panel or a COM port on a personal computer. Connect the LINE RJ11 jack to an analog telephone line. Connect the PHONE RJ11 jack to an analog telephone set (optional). Connect the external power supply to the POWER SUPPLY INPUT jack of the EF2241. Set the country code on the EF2241. By default the phone interface will be disabled until you select a country code for the telephony interface. This can either be done with the front panel LCD menu, or the RS-232 interface. The country code only needs to be selected the first time or when the country of the installation is changed.

DEVICE IDS ON THE EF BUS

When considering which Device IDs can be used for which Vortex device, decide how many devices have the ability to transmit on the W, X, Y, and Z busses, and how many have the ability to transmit on the P Bus. The EF2280, for example can only transmit on the W, X, Y, and Z busses while the EF2241 can transmit on the W, X, Y, and Z busses as well as the P bus. Up to 8 devices can transmit on the W, X, Y, and Z busses. Similarly, up to 8 devices can transmit on the P bus. Note that the EF2241 counts as one of both types. All devices that can transmit on W, X, Y, and Z must have unique device IDs. Vortex devices of the same type must have unique device IDs. If the device IDs of linked Vortex devices are the same, the front panel LCD screen will display EFBus Error: Dev. ID Conflict.
CONNECTING MULTIPLE VORTEX DEVICES
Up to 8 Vortex devices in combination can be linked together at one time (See Device IDs on the EF Bus above). Each unit in the chain must have a unique Device ID. Use the EF Bus to link multiple Vortex devices together. The following steps should be followed to connect the EF Bus: 1. 2. 3. 4. Set a unique Device ID for each Vortex device. The Device IDs range from 00 to 07. Power off all units. Connect the RS-232 remote control device to any Vortex device in the chain. Connect the provided Cat-5 cable between the EF BUS OUT of the first device, and the EF BUS IN of the second device.

AGC Level Meter

Read AGC level

Adjust Input Gain here

Figure 9. AGC Meter on the Diagnostics page of the Conference Composer software. The gain adjustment is the same gain control on the Mic/Line Inputs page in Conference Composer.
IF THE AGC METER SHOWS. positive gain negative gain an average level of 0 dB
ADJUST THE INPUT GAIN IN THIS WAY. Increase gain by the level shown in the box. Decrease gain by the level shown in the box. Youve set the Input Gain to a good level!
Table 1: How to set the Input Gain using the AGC meter on the Conference Composer Diagnostics page.
Set Levels on Line Input Channels
Set the line input channel gains (Channels A-D) to match the nominal level of the incoming equipment. The line inputs have a maximum nominal level of 0 dBu. If your incoming line level inputs have a higher nominal level than 0 dBu you will want to use a pad to remove the level. If you are connecting from equipment that has RCA plugs, you will most likely need 8 dB of gain on the Input and -8 dB of gain on the Output.
INTEGRATING Customize Setting for Your Particular Application
The following sections will describe customizing parameters on the EF2241 for your particular application if you are not using Preset 0.
SET UP THE PHONE INTERFACE
The echo canceller reference of the EF2241 is by default set up with the phone input already assigned to it. By default the phone interface will be disabled until you select a country code for the telephony interface. This can either be done with the front panel LCD menu, or the RS-232 interface. The country code only needs to be selected the first time or when the country of the installation is changed. Send audio from the phone to the outputs on the same device by unmuting the crosspoint of Input T on that output. Send audio from the phone to other linked devices by using the P bus output in the EF Bus. The linked devices can take the phone signal off the bus from the P Bus input.
BUILD YOUR ECHO CANCELLER REFERENCE
The acoustic echo canceller (AEC) reference should generally contain exactly the same audio signals as what is coming out of your loudspeaker(s), since the signal output from the loudspeaker is what is then picked up by the rooms microphones causing acoustic echo. Note that this statement is a general statement. Conditions for this being true follow:

Operating as an Independent Automixer Operating as One Automixer with Multiple Vortex Devices Default Settings
To set the Vortex device to operate as an independent automixer (or two), set the BUS MIXER parameter to 0, or to a number that is different from any other automixer group on the EF Bus.
To set the Vortex device to operate as one automixer across several devices, set the BUS MIXER parameter on all devices to the same automixer group.
In the default preset (Preset 0), the Bus Mixer for Automixer 1 is set to 1 and the Bus Mixer for Automixer 2 is set to 2. This means that in the default mode, all linked devices will work together as one complete automixer.
CONFIGURE THE MATRIX MIXER
The matrix mixer allows arbitrary crosspoint gains in 1 dB increments between any input and output signal. If you have a PC, use the Conference Composer software (See the Conference Composer User Guide) to set the matrix parameters. If you would like to set matrix parameters from the LCD menus, see Matrix Menu on page 34 for descriptions and instructions on setting matrix mixer parameters.
BUILDING YOUR SYSTEM WITH MULTIPLE VORTEX DEVICES
The following is a checklist for building a system with mutliple devices: 1. 2. 3. 4. Assign Inputs. Assign Outputs. Configure submatrix (the EF Bus). Configure your echo canceller reference.
INTEGRATING 1. Assign Inputs
Assign each audio source to an input. Remember to include the conferencing equipment such as a video codec and any program audio.

2. Assign Outputs

Try to assign as many outputs as you can to one Vortex device to make a simpler submatrix. Remember that Outputs 1-4 can also be used as outputs of the matrix. The bussing gets more complicated if you choose to spread your outputs over several units.
3. Configure the submatrix.
To link multiple devices together, use the submatrix on the EF Bus to configure which signals to receive from other devices that have put their signals on the Bus.

THE EF BUS

The EF Bus is a high speed, low delay digital bus that includes the W, X, Y, and Z audio busses, the P bus, as well as the echo canceller bus reference and remote control information (for other EF devices). It can link up to 8 Vortex devices. The W, X, Y, and Z busses include NOM information and can be used for sharing microphone inputs, or for sharing mono or stereo program information.On the EF Bus page in Conference Composer, the inputs coming in to each submatrix labelled WB0, WB1,. WB7 correspond with the device ID of the bus that is transmitting. The B denotes Bus. The submixes themselves, denoted as WM0, WM1, and WM2 are mixes that are input into the main matrix. The M denotes Mix.

PRESETS

After configuring your Vortex device, save your settings to a User Preset (PRESETS 16-47). Also, set the POWER ON PRESET to the User Preset you have saved to. The POWER ON PRESET determines how the unit is configured upon power up. If you have multiple Vortex devices in your system, save to a User Preset on each unit and set the POWER ON PRESET accordingly.

OTHER EF2241 FEATURES

For information on Macros, Logic Inputs, Logic Outputs, Input Filters and Output Filters, please refer to the Conference Composer User Guide.

LCD MENU STRUCTURE

LCD Menu Tree
The EF2241 LCD menu structure is made up of nine menu trees: SYSTEM, PHONE CONTROL, INPUTS, OUTPUTS, AUTOMIXER, MATRIX, PARAMETRIC EQ, PRESETS, and MACROS. Each menu tree is organized by levels and branches into multiple subcategories. The branches end with an adjustable parameter or value.

EF2241 System Settings

System Power On Preset Acknowledgement Mode Bus Reference Error Messages Front Panel Lock Front Panel Password Device ID Baud Rate Flow Control Meter Non-Volatile Memory Lock Non-Volatile Memory Password Screen Saver Last Restored Preset Software Version Reboot Vortex
Phone Control Auto Answer Auto Hangup Call Progress Auto Hangup Loop Drop Country Select Dial Tone Gain DTMF Gain Ring Tone Enable
Inputs 1-4, T, A-D Acoustic Echo Cancellation (In 1-4) Automatic Gain Control (AGC), In 1-4, T AGC Maximum (In 1-4, T) AGC Minimum (In 1-4, T) Gain Level Mic/Line (In 1-4) Mute Noise Cancellation (In 1-4, T) Noise Cancellation Level (In 1-4, T) Phantom Power (In 1-4) Reference Assign (In 1-4) AEC Suppression (In 1-4) Line Echo Cancellation (In T) Dynamics Processing (In T) LEC Suppression (In T)
Outputs Output Gain Mute Output NOM Attenuation On/Off Output Delay Enable Output Delay Dynamics Processing (Out T)
Automixer Global Settings Decay Hold Time Camera Gating Threshold Mixer Settings (AM1/AM2) Bus ID Chairman Mode Chairman Mic Last Mic On Mode Last Mic Number Local Max NOM Global Max NOM Off Attenuation Automixer Reference Reference Bias
Matrix Main Matrix Gain Gate (In 1-4) Mute SubMatrix Gain Mute
Parametric EQ Input/Output EQ Channel Band Filter Type Frequency Bandwidth Gain Slope Filter Enable
Presets Restore Save Presets 16-47 Delete Presets 16-47

Macros Run Macro (0-255)

Channel Settings (Inputs 1-4) Automixer Threshold Type Gating Mode Adaptive Threshold Manual Threshold Gate Priority
Figure 16. LCD Menu Tree. DOWN Scrolls backward through menu items at particular level or decreases the value of a parameter UP Scrolls forward through menu items at particular level or increases the value of a parameter. ENTER Enters the menu and allows you to select and change parameter values. ESC Returns to the top of the next highest level of menus HOME Returns to the top of the menu structure.

Error Messages. Turns error messages On or Off. Front Panel Lock. Locks or unlocks the front panel. When the front panel is
locked, you can see the parameters but you cannot change them. The default passcode is aspi (case is important).
Front Panel Passcode. Once the device has been unlocked, the passcode may
be changed. At the FRONT PANEL PASSCODE menu, press ENTER and then enter a passcode and press ENTER until you reach the end of the screen.
Device ID. Selects the Device ID of the unit. Baud Rate. Selects baud rate of the RS-232. Flow Control. Selects flow control between Hardware, None, or Auto. LCD Contrast. Controls the contrast level of the front panel liquid crystal display
(LCD). Higher numbers result in darker characters on the display, lower numbers result in lighter characters.
Meter. Selects which signal is displayed on the front panel LED meter. Non-Volatile Memory Lock. Controls the non-volatile lock feature. When
the non-volatile memory is locked, you can query the settings but will get an error if you try to change them.
Non-Volatile Memory Password. This feature sets or queries the non-volaORY
Figure 17. EF2241 System submenu
tile lock password. This password is used in conjunction with NON-VOLATILE MEMLOCK. The default password is aspi (case is important).
Screen Saver. Enables or disables the screen saver on the LCD panel. You can

also set the idle time.

Last Restored Preset. Displays the last restored Preset. Software Version. Queries the software version. Reboot EF2241. Cycles power on the EF2241.

PHONE CONTROL

Auto Answer. Enables or disables auto answer. Auto Hangup Call Progress. Enables or disables auto hangup based on call

progress messages.

Auto Hangup Loop Drop. Enables or disables auto hangup cased on loop
Country Select. Selects the country where you are using the unit. This must be set before you can use the unit and does not need to be set again unless you use the unit in a different country. Dial Tone Gain. Adjusts the gain of the dial tone. DTMF Gain. Adjusts the gain of the DTMF tones. Ring Tone Enable. Enables or disables ring tones.
Figure 18. EF2241 Phone Control submenu

(LEC).

LCD MENU STRUCTURE Phone Input Gain. This parameter adjusts the gain level of the Phone input. Phone AGC. Enables or disables automatic gain control on the Phone.

OUTPUTS

The OUTPUT menu contains GAIN, NOM ACTIVE, and MUTE. As with the INPUT menus, this is done on a per channel basis.

Outputs

Output Gain Mute Output NOM Attenuation On/Off Output Delay Enable Output Delay Dynamics Processing
Output Gain. Choose the gain applied to each output signal using this parameter. The default setting is 0 dBu. Though the EF2241 allows for positive output gain, you should always try to adjust input gains to a good level so that the output gain is 0 dB. If you find that you need a positive output gain from the EF2241, first check your input gain to make sure you are getting a good level (around 0dB). Keep the output gain at around 0 dBu. Then, for the best gain structure, use your amplifier to raise the volume in your system. Mute Output. Use this to mute or unmute each Output. NOM Active. This allows you to select whether the NOM attenuator is active for a particular output channel (Outputs 1-4, A-D). The NOM attenuator will attenuate the output signal by 10*log10(NOM) where NOM is the number of open microphones in that particular output channel. NOM is calculated based on the number of open microphones for each signal that is in the output. Output Delay Enable. This allows you to enable delay to each of the outputs. Output Delay. Sets the amount of delay on the output. The default value is 0 ms.
The range is 0 to 340.0 ms in 0.1 ms increments.
Figure 20. EF2241 Outputs submenu
Dynamics Processing. Enables or disables compression on Output T.

AUTOMIXER MENU

These parameters configure how the Vortex device automatic microphone mixer operates. Parameters include the following: DECAY TIME, HOLD TIME, CAMERA GATING THRESHOLD, BUS ID, CHAIRMAN MODE, CHAIRMAN MIC, LAST MIC ON MODE, LAST MIC NUMBER, LOCAL MAX NOM, GLOBAL MAX NOM, OFF ATTENUATION, AUTOMIXER, THRESHOLD TYPE, GATING MODE, ADAPTIVE
Automixer Global Settings Mixer Settings Channel Settings
Figure 21. EF2241 Automixer submenu
THRESHOLD, MANUAL THRESHOLD, and GATE PRIORITY.
Figure 22. Automixer parameters.
Automixer Global Settings

Decay Hold Time Camera Gating Threshold
LCD MENU STRUCTURE Mixer Settings.
Bus Mixer. This command is used to assign one of the two internal automixers to one of the EF Bus automixer groups. For example, consider three Vortex devices each of which has 2 microphones assigned to Automixer 1 and 2 microphones assigned to Automixer 2. Now, if each of these devices sets their Automixer 1 to have Bus Mixer 5, then the three automixers (one from each device) will work as a single automixer containing 6 (3 x 2) microphones. Setting Bus Mixer to 0 means that the automixer is not grouped on the EF Bus. Chairman Mode. Enables or disables Chairman Mode for the specified automixer. Chairman Mic. Sets the Chairman Microphone for the specified automixer. Last Mic On Mode. Sets Last Mic On mode for the specified automixer. Last Mic Number. Sets the microphone number that will remain on when Last Mic On mode is set to manual. Setting this value to 0 will cause the automixer to leave the last open microphone on. The last microphone number is specified for each automixer, but is only used in manual Last Mic On mode. Local Max NOM. Sets the maximum number of open microphones (NOM) limit for the specified automixer. This NOM limit is a local limit, meaning that this limit applies only to the specific Vortex device that it is set on. Global Max NOM. Sets the global maximum number of open microphones (NOM) limit for each linked automixer. The maximum value for this command is 64. This NOM limit is a global limit, meaning that this limit applies to all linked automixers with the same Bus ID. Off Attenuation. Sets the Off Attenuation (in dB) for the specified automixer. Setting this value to 18 would result in the microphone signals being attenuated by 18 dB when gated off. This value is set independently for each of the automixers. Automixer Reference. When enabled, the echo canceller reference becomes a muted input in the automix so that far end audio coming from the speakers does not gate on local microphones. Reference Bias. Adjusts how much gain is applied to the Automixer Reference signal. The higher the gain, the harder it will for local talkers to gate on a microphone.

Automixer

Mixer Settings (AM1/AM2) Bus ID Chairman Mode Chairman Mic Last Mic On Mode Last Mic Number Local Max NOM Global Max NOM Off Attenuation Automixer Reference Reference Bias
Automixer (Inputs 1-4). This allows you to select which automatic microphone mixer (1 or 2) a particular microphone channel is assigned to. A microphone may only be assigned to automatic mixer 1, automatic mixer 2, or neither (but not both). Threshold Type. Sets automatic (also referred to as adaptive) or manual automatic gating thresholds per input. Gating Mode. Sets the automixer gating control mode for specified input channel. The possible modes are Normal Gating, Microphone Forced On, or Microphone Forced Off. Adaptive Threshold. This is also referred to as the Adaptive Threshold. This allows you to determine when to gate a microphone on based on an estimate of the background noise level. The default value is to gate a channel on if it is more than 10 dB louder than the background noise level. Values range from 0 to 100 dB. To set the adaptive threshold, scroll through the adaptive threshold range and select the desired adaptive threshold by pressing ENTER. Manual Threshold. Sets the automixer gating threshold for the specified input channel. This value is only used if the input set to Manual Gating via the THRESHOLD TYPE option.

Filter Type Frequency Bandwidth Gain Slope Filter Enable
Figure 24. EF2241 Parametric EQ submenu
Restore. Restores the selected preset. Save. Saves the selected user preset (Presets 16-47). Factory presets (Preset 0-15)

cannot be overwritten.

Presets
Restore Save Presets 16-47 Delete Presets 16-47
Delete. Deletes the selected user preset (Presets 16-47). Factory Presets (Presets 015) cannot be deleted.
Figure 25. EF2241 Presets submenu

MACROS

Run Macro (0-255). Allows you to run macros from the front panel menu.
Figure 26. EF2241 Macros submenu

TROUBLESHOOTING

AUTOMATIC MICROPHONE MIXER
No microphones are gating Some microphones are not gating
Check if the microphones are muted. Are microphones part of one of the 2 automixers?
Check if the microphones are assigned to an automixer. Check if the microphones are muted. Check microphone levels. Are microphones set to the appropriate mic or line level? Is phantom power on where needed? The Hold Time may be too low. Microphone channels gating On and Off too frequently during short pauses in speech might be the result of setting the Hold Time too low. Check Gating settings. Are microphones Forced Off? Is Chairman Mode on? If you have assigned a Chairman Mic, all other microphones will gate Off once this microphone gates on. Check Gating Priority. If your inputs have a Gating Priority of 4, the microphones may not gate as frequently. Check Maximum Number of Open Microphones. This parameter sets the number of open microphones allowed at any time. If this parameter is set too low, the microphones may not gate as often as you wish. Adjust the Adaptive Threshold if the Gate Threshold is set to Adaptive or adjust the Manual Threshold if the Gate Threshold is set to Manual. For Adaptive Gate Threshold, set the Adaptive Threshold lower so that the microphone will gate On when lower level signals are present at the microphone. For Manual Gate Threshold, set the Manual Threshold to a lower absolute threshold.

talker to the microphone. The microphone audio will most likely also be muddy and reverberant. The installer or user may try to solve these microphone audio quality problems by turning up the microphone amplification, thus adding to the room gain. This problem can be remedied by proper microphone selection (pickup pattern, directionality) and placement, coupled with proper microphone input calibration. 3. A third common cause of room gain problems is excessive coupling between loudspeaker audio and microphones. This can be addressed by reducing the microphone coupling, either by positioning microphones so that their pickup patterns are biased away from the loudspeaker audio (and direct reflections of loudspeaker audio), repositioning loudspeakers, or reducing the loudspeaker amplification. In summary, any amplification applied between the reference input and the microphone inputs can add to room gain problems. To avoid problems, ensure that the Reference input signal is not too low, and the microphone input signals are not too high. Run the built-in EF2241 Room Gain test to verify that you do not have room gain problems (See Verify Room Gain on page 14). COMMON CAUSES OF EXCESSIVE ROOM GAIN Excessive remote (reference) input amplification Excessive microphone amplification REMEDY Apply enough gain to the codec, phone or program audio inputs which will make up the Reference input signal. Select proper microphones for talker distance according to pickup pattern and directionality and properly calibrate mic inputs. Reduce mic coupling by repositioning mics or loudspeakers, or by reducing loudspeaker amplification.
Excessive coupling between loudspeaker audio and microphones
Table 3: Summary of Excessive Room Gain.
In-Conference Quick Check
If you experience residual echo problems during a conference, you can quickly check that the reference and microphone levels are calibrated and not causing room gain problems by using the Room Gain parameter (See Verify Room Gain on page 14). If this excessive coupling activity level is evident on only one microphone input channel, that microphone channel should either be redirected to reduce coupling to loudspeaker audio, or recalibrated as it will need to be turned down. If the excessive coupling activity is observed on all (or most) microphone channels, then this indicates either that the room audio is too loud or the reference signal may need to be recalibrated (this will be indicated by observing low activity levels on the SIGNAL LEVEL METER).

Excessive Microphone Amplification
For the EF2241 to adapt effectively, saturation (overload or clipping) must not occur at the A/D converter supplying the microphone input. Saturation introduces nonlinear signal distortions into what the AEC expects is a linearly echoed version of the remote speech.
Nonlinear distortion causes a degradation or divergence of the AECs internal model of the room acoustics. In this situation, the EF2241 cannot effectively cancel room echoes and a substantial amount of echo may be heard by the remote party. Excessive microphone amplification also increases room gain (See Excessive Room Gain on page 40.). You can check for excessive microphone amplification by observing the front panel LEVEL INDICATOR during a normal conference. The first yellow LED should illuminate frequently. If the second yellow LED is illuminated constantly during normal speech or if the red LED illuminates or even flickers, reduce the microphone input level.
Note. Before you readjust the microphone input levels, check to
make sure you are looking at the correct channel on the LEVEL INDICATOR.
Note. If you adjust the MIC/LINE INPUT level, you will affect the
room gain. Check to make sure that the room gain limit is not exceeded. See Verify Room Gain on page 14.
Insufficient Microphone Amplification
Grossly insufficient microphone gain degrades EF2241 performance and weakens the out-bound speech power level. This has the effect of reducing the signal-to-noise ratio of the microphone signal, which is analogous to raising the background noise level in the room. Because this noise is uncorrelated with the echoes within the room, the EF2241s ability to adapt and cancel echoes will be less than optimal. A second effect of insufficient microphone gain is that the power of the microphone input signal may be substantially lower than that of the remote input signal. This reduces the ability of the decision logic to determine whether the AEC should be in transmit, receive, or double-talk mode. This effect may reduce the effectiveness of the EF2241 in canceling echoes. You can check for insufficient microphone amplification by observing the front panel LEVEL INDICATOR during normal conferencing conversation. The first yellow LED should illuminate frequently. If the LEVEL INDICATOR never illuminates beyond one or two green LEDs during normal speech, increase the microphones input level.
room gain. Check to make sure that the room gain limit is not exceeded. See Room Gain on page 39.

COMPLIANCE

The Vortex EF2241 complies with the ITU G.167 Recommendation for AEC, FCC part 15, and CE requirements.

USA and Canada

This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: 1. 2. This device may not cause harmful interference, and This device must accept any interference received, including interference that may cause undesired operation.
This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference in which case the user will be required to correct the interference at his own expense. In accordance with part 15 of the FCC rules, the user is cautioned that any changes or modifications not expressly approved by Polycom Inc. could void the users authority to operate the equipment. This Class [A] digital apparatus complies with Canadian ICES-003. Cet appareil numrique de la classe [A] est conforme la norme NMB-003 du Canada.

US Telco requirements

This equipment complies with part 68 of the FCC Rules. Please refer to the labeling on equipment for the following information: Registration Number Ringer Equivalence Grantee's Name Model Number Serial Number and/or Date of Manufacture Country of Origin
If requested this information must be provided to the telephone company Notes This registered equipment may not be used with party lines or coin lines. If trouble is experienced the customer shall disconnect the registered equipment from the telephone line to determine if the registered equipment is malfunctioning and that if the registered equipment is malfunctioning, the use of such equipment shall be discontinued until the problem has been corrected.
If, in the unlikely event that this equipment causes harm to the network, the telephone company will notify you in advance that temporary discontinuance of service may be required. But if advance notice isn't practical, the telephone company will notify you as soon as possible. Also, you will be advised of your right to file a complaint with the FCC if you believe it necessary. The telephone company may make changes to its facilities, equipment, operations or procedures that could affect the operation of the equipment. If this happens the telephone company will provide advance notice so you can make the necessary modifications to maintain uninterrupted service.

doc1

Polycom Vortex EF2241
Acoustic echo/noise canceller with automatic microphone/matrix mixer and phone add
Benefits Industry leading audio quality Design enhancements improve the audio experience for even the most demanding listening environments; support for the broadest range of microphones in the industry Versatile telephone connection With the Vortex EF2241, add standard (PSTNbased) calls or enjoy wideband audio performance with the Polycom SoundStation VTX 1000 conference phone Hear every word with Polycoms industry-leading acoustic echo and noise cancellation Ensures the highest possible audio quality between sites with 20 kHz bandwidth, 270 ms tail length, more than 10 dB of room gain, and a convergence rate of over 40 dB per second on each mic/ line channel Integrated voice and video solution May be linked with additional Vortex Installed Voice products including Vortex EF2201, Vortex EF2210, Vortex EF2211, Vortex EF2241 and Vortex EF2280; Polycoms SoundStation VTX 1000 wideband conference phone and Polycoms video conferencing systems including the VSX 7000 and 8000 product lines Easy setup Conference Composer and Polycom InstantDesigner software make configuration fast and simple Easy to operate Controllable via Crestron, AMX or other room control systems, RS-232, the Vortex EF-IR11 infrared controller, or from the units front panel; 16 factory presets and 32 user configurable presets allow quick selection of operating parameters Adaptable to user requirements Userprogrammable delays on all outputs, 5-band parametric EQ on all inputs and outputs, intelligent Automatic Gain Control, 256 macros, system diagnostics, and other tools ensure the best performance for all room conditions
Expand the audio capabilities of your installed conferencing facility and easily add phone calls.
The Vortex EF2241 enhances the productivity of conferencing and distance learning installations through industry-leading acoustic echo and noise cancellation, plus a DSP-based telephone interface - with support for up to four microphones. Designed as a companion to Polycom's Vortex EF2280 model, as well as for standalone applications, the Vortex EF2241 automatically mixes four microphones and other audio, while canceling acoustic echoes and annoying background noise. The Vortex EF2241 includes an interface to the standard (PSTN) telephone network. It provides connections for multiple microphones and auxiliary inputs and includes a 10 Watt power amplifier for driving loudspeakers in the room. The Vortex EF2241 supports four auxiliary audio sources and automatic mixing of up to four microphones per device. If more microphones or telephone interface channels are needed, additional Vortex units can be linked. Each microphone channel features industry-leading acoustic echo cancellation to prevent received signals from being retransmitted to their original locations. Neural network Automatic Gain Control (AGC) reacts only to valid speech patterns, bringing voices within desired levels. User-adjustable AGC controls for each channel, and settings for the five-band parametric EQ, are offered on all input and output channels. The high-quality audio choice to access the power of Polycom unified collaborative communications solutions. With the greatest breadth and depth of integrated video, voice, and Web solutions, only Polycom delivers the ultimate communications experience. Our market-leading conferencing and collaboration technologies, supported by world-class service, enable people and organizations to maximize their effectiveness and productivity. Add to that the most experience and proven best-practices in the industry, and its clear why Polycom has become the smart choice for organizations seeking a strategic advantage in a real-time world.
Connect. Any Way You Want.
Polycom Vortex EF2241 Specifications
System Block Diagram and Rear Panel
Output 1 Output 2 Output 3 Output 4 Output A Output B Output C Output D Amplifier Output

Tone Gen. Phone D/A

Matrixing

LEC A/D

Telephone Line

PEQ EN EN EN EN AGC

PEQ AGC

PEQ TLP AGC

Level adjust

RS-232 EF Bus

Phantom Mic/Line

Phantom

Mic/Line

Control/Status

Mic/Line 1
Mic/Line 2 Mic/Line 3 Mic/Line 4

Input A

Input B

Input C

Input D
EN (Echo Canceller, Noise Canceller)
Ungated Echo Canceller R1, R2, Bus Noise Canceller On Off AGC On Off PEQ Mute On Off Gate Gated
TLP (Telephone Line Processing)

AGC On Off

Noise Canceller Parameters
Call Prog Detect Parameters

Mute On Off

Parameters
Dimensions 19 (483 mm) W x 9.6 (244 mm) L x 1.75 (45 mm) H (one rack unit) Weight 4 lbs. (1.8 kg) dry, 5.5 lbs. (2.5 kg) shipping Connectors RS-232: DB9F EF Bus In/Out: RJ45 Control /Status: DB25F Telephone Line/Set: RJ11 Audio: Mini (3.5mm) quick connect terminal blocks
Outputs Output Gain: -100 to 20 dBu in 1 dB steps, software adjustable Maximum output amplitude: +23 dBu, 1% THD + N Nominal output level: 0 dBu (0.775 V rms) Output impedance: 33 Ohm, each leg to ground (drives 600 Ohms) Output EMI Filter: Pi filter on all audio outputs Amplifier Maximum Output Power: 8 Ohm, >10 W, 20 - 20,000 Hz, + 0.2 /- 0.5 dB Signal to noise ratio: 8 Ohm, >90 dB "A" weighted, 20 - 20,000 Hz Impedance: 4-16 Ohm Telephone Input gain: -100 to +20 dB in 1 dB steps, software adjustable Nominal transmit level: 0 dBu in Vortex yields -15 to -17 dBm to phone (country code dependent) Off hook loop current: 16 mA (minimum) to 120 mA (maximum) Output gain: -100 to +20 dB in 1 dB steps, sofware adjustable Frequency response: 250-3300 Hz Dynamic range: >70 dB FS, 250-3300 Hz, "A" weighted
Power External supply (provided) Input voltage of 110-240 VAC; 47-63 Hz; power consumption 30 W Inputs Phantom power: 48 V DC, software selectable Analog input Gain: 0 to 30 dB Mic/line inputs in 1 dB steps, software adjustable Mic/Line switch gain: 33 dB Maximum input amplitude: +19 dBu, 1% THD + N Nominal level: 0 dBu (0.775 V rms) Equivalent input noise: <-124 dBu, 20 - 20,000 Hz Input Impedance: 10 kOhms Input EMI Filter: Pi filter on all audio inputs
System* Frequency response: 20-20,000 Hz, + 0.2 /- 0.3 dB Idle channel noise: <-100 dB FS "A" weighted, 20 - 20,000 Hz, 0 dB gain Dynamic range: >100 dB FS "A" weighted, 20 - 20,000 Hz, 0 dB gain Linearity: 0 dB FS to -110 dB FS +/- 1 dB THD+N: < -90 dB FS Common Mode Rejection Ratio: <-61 dB, 20 - 20,000 Hz, no weighting Cross talk: <-104 dB, 20-20,000 Hz, channel-to-channel Latency: Mic/Line inputs to outputs: 13 ms, processing enabled Acoustic Echo Cancellation Span: 270 ms Total Cancellation: >65 dB Convergence Rate: 40 dB/second Noise cancellation: 0-15 dB, software selectable Operating Temperature: 0 - 40 C Control Inputs: Contact closure Status Outputs: 5 V, 20 mA each Line Echo cancellation (telephone hybrid) - Echo cancellation: 40 dB, total > 60 dB - Convergence Rate: 30 dB/second - Cancellation span: 32 ms *Unless noted, all values are valid for all channels at line level

2004 Polycom, Inc. All rights reserved.
Polycom, the Polycom logo, and Vortex are registered trademarks and SoundStation VTX 1000, VSX, Conference Composer and InstantDesigner are trademarks of Polycom, Inc. in the U.S. and various countries. All other trademarks are the property of their respective companies. Specifications subject to change without notice. For Architect's and Engineer's Specifications, please visit Polycom.com, or contact the Installed Voice Business Group at 1.770.641.4400.

Polycom Headquarters:

www.polycom.com
4750 Willow Road, Pleasanton, CA 94588 (T) 1.800.POLYCOM (765.9266) for North America only. For North America, Latin America and Caribbean (T) +1.925.924.6000, (F) +1.925.924.Bath Road, Slough, Berkshire SL1 4DX, (T) +44 (0)1753 723000, (F) +44 (0)Polycom Hong Kong Ltd., Rm 1101 MassMutual Tower, 38 Gloucester Road, Wanchai, Hong Kong, (T) +852.2861.3113, (F)+852.2866.8028
Part No. 3726-82241-001 Rev. 10/04
Polycom EMEA: Polycom Asia Pacific:

 

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