449 lines
26 KiB
HTML
449 lines
26 KiB
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<!doctype HTML public "-//W3C//DTD HTML 4.0 Frameset//EN">
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<html>
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<meta http-equiv="Content-Type" content="text/html; charset=windows-1252">
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<meta name="generator" content="Adobe RoboHelp 9">
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<title>AEC Cards</title>
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<link rel="StyleSheet" href="HtmlHelp.css" type="text/css">
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<style title="hcp" type="text/css">
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</style>
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</head>
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<body>
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<h1 class="firstHeading">AEC Cards</h1>
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<p>There are 2 AEC versions available, the original AEC algorithm or full
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bandwidth AEC.  The type can be selected in properties for the AEC
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card or AEC processing object.  The full bandwidth AEC option will
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give improved audio performance and we would recommend that you enable
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this option.  Any designs which have been created in an earlier version
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of London Architect where this feature was not available will retain the
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original AEC when loaded.  Existing systems may be upgraded to the
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full bandwidth AEC version but additional system tuning may be required
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for optimum performance. New designs will default to the full bandwidth
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AEC.</p>
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<h2>Front Panel Information on Soundweb London Devices</h2>
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<p>AEC cards can only operate within a BLU-800/BLU-805/BLU-806,
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BLU-160, BLU-320/BLU-325/BLU-326
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or  BLU-120 at a sample rate of 48kHz. If an AEC card is fitted to
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one of these devices which had already been configured for 96kHz the AEC
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LEDs on the front panel will flash to indicate this. Attempts to go online
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will fail and be recorded in the Event Log. It will be necessary to change
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the sample rate in the design to 48kHz before going online to the device.</p>
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<h2>AEC Definitions</h2>
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<span class="hcp1">Convergence Rate</span>
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<br>Measures the speed of the linear processing component
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of the AEC algorithm and does not include the non-linear processing or
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suppression (NLP) as dictated by industry standards. This means this is
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a measure of how fast the algorithm can recognize and remove echo from
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the signal path.
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<br> 
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<br><span class="hcp1">Double Talk</span>
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<br>Both far and near side speech are present.
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<br> 
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<br><span class="hcp1">Echo Return Loss (ERL)</span>
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<br>This is a measure of the coupling between the AEC reference
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signal and the AEC input signal.
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<br> 
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<br><span class="hcp1">Echo Return Loss Enhancement
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(ERLE)</span>
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<br>This shows the loss through the linear AEC algorithm
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(not including the non-linear processing.)
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<br> 
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<br><span class="hcp1">Far Side (Reference)</span>
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<br>This is the remote side of the conference which will
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be heard from the near-side speakers.
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<br> 
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<br><span class="hcp1">Gain Structure </span>
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<br>Proper gain structure will provide an adequate signal
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to noise ratio and reasonable headroom for an input signal.
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<br> 
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<br><span class="hcp1">Near Side (Local)</span>
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<br>This is the local side of the conference where the echo
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canceller is located.
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<br> 
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<br><span class="hcp1">Non-Linear Processing
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(NLP)</span>
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<br>The non-linear processing increases the power of the
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echo cancellation for difficult acoustic environments.
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<br> 
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<br><span class="hcp1">Noise Cancelation (NC)</span>
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<br>Noise cancellation removes ambient noise from the AEC
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signal (e.g. computer fan noise).
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<br> 
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<br><span class="hcp1">Voice Activity Detection
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(VAD)</span>
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<br>Detects whether the audio is speech or silence/background
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noise.
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<h2>Acoustic Echo Cancellation (AEC)</h2>
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<p>Acoustic echo occurs in a conferencing system when the far-side speech
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played in the loudspeakers is picked up by microphones in the room and
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is transmitted back to the far side. This transmitted signal is a delayed
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version of the original, which causes the echo.</p>
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<p><img src="AEC Process.PNG" alt="" width="445" height="166" border="0" class="hcp2"></p>
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<p> The received far-side signal does not transfer directly from the
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speaker to the microphone, but is subject to the artifacts of the room.
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This may include differing signal paths causing reverb, frequency filtering
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and attenuation. These effects are the transfer function of the room.
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The transfer function of the room is also dynamic, as objects in the room
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move or the microphone moves position.</p>
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<p>To correctly subtract the required signal, the AEC therefore needs to
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simulate the dynamic room transfer function. It can then apply that transfer
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function to the received signal and correctly subtract the modified original
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signal.</p>
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<p>Each AEC card consists of 4 AEC input channels. Each channel offers
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the following features:</p>
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<ul>
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<li>Independent 20Hz - 8kHz algorithm for the original AEC or 20Hz
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- 20kHz algorithm for the full bandwidth AEC selectable</li>
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<li>Individual AEC references</li>
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<li>Automatic Gain Control (AGC)</li>
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<li>Noise Cancellation (NC)</li>
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<li>Adaptive (Speech Passing) Non-Linear Processing (NLP)</li>
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<li>Extremely fast convergence rates of 49dB/s</li>
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</ul>
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<p class="hcp3">NOTE: AEC Input cards can only be used in
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BLU-800/BLU-805/BLU-806, BLU-320/BLU-325/BLU-326, BLU-160 or BLU-120 devices.</p>
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<h2>AEC Card Control Panel</h2>
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<p>The AEC default control panel is ordered in two groups of controls for
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every input channel. The first group of controls are identical to the
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standard Soundweb London input cards and function in the same manner.
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These controls are the audio input meter (configurable as Pre or Post-AEC),
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input meter controls - Attack, Release, Reference, and Phantom Power -
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for each input channel. The second group of controls are the AEC controls.</p>
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<h4>AEC Control Panel</h4>
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<p>The basic AEC control panel allows enabling and disabling of AEC and
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AGC, and allows setting levels for noise cancellation, non-linear processing
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and signal threshold.  The AEC control panel for the <a href="BLU-101_Fixed_card_configuration_with_BLU_link.htm">BLU-101</a>
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and <a href="BLU-102_Fixed_card_configuration_with_BLU_link.htm">BLU-102</a>
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<a href="Conferencing.htm">AEC processing object</a> differs from that
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of the AEC input card since it does not show the analogue input meters
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and input controls which can be found on the control panel for the analogue
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input card for those devices.</p>
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<p> <img src="AEC_control_panel.JPG" alt="" width="832" height="697" border="0" class="hcp2"></p>
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<p> </p>
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<h4>AEC</h4>
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<p>This button enables or disables AEC processing for each channel. When
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this button is enabled the AEC algorithm will remove the acoustic echo
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from the audio channel with linear processing and with a specified amount
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of non-linear processing. (See <a href="AEC_Cards.htm#NLP_Level">NLP</a> Level below.)</p>
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<a name="ERL_Meter" id="ERL_Meter"></a>
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<h4>ERL Meter</h4>
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<p>The Echo Return Loss (ERL) meter is a measure of the room's natural
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attenuation of the far-side audio as it leaves the speaker(s) and re-enters
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the microphone(s). This parameter is controlled by proper gain structure
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setup (ensuring a good signal to noise ratio and reasonable headroom for
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the AEC input signal). A proper gain structure is critical for distortion
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free sound and optimal performance for AEC. This is the <b>single most
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important parameter</b> when setting up the AEC system.</p>
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<p>The AEC algorithm will only be able to recognize and remove echo to
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its best extent when this meter is displaying in the 'green' range. The
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'green' range is indicated on the control panel below 0dB. The algorithm
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will continue to converge over 0dB, but the convergence rate will decrease
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in that range.  If the ERL meter is over 10dB, convergence should
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not be expected.</p>
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<p>This meter will not update during double-talk. It is updated based on
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far-side speech only.</p>
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<a name="ERLE_Meter" id="ERLE_Meter"></a>
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<h4>ERLE Meter</h4>
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<p>The Echo Return Loss Enhancement (ERLE) Meter measures how much acoustic
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echo is being removed from the signal path. This measurement consists
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of the natural room attenuation as indicated by the ERL meter and the
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amount of echo removed by the AEC algorithm. A lower signal indicates
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more echo being removed. The lower the meter, the better.</p>
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<p class="hcp3">NOTE: As dictated by industry standards,
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NLP contributions are not included in this reading. NLP contributions
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are made <span class="hcp1">in addition to</span> this meter's
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reading.</p>
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<a name="NLP_Level" id="NLP_Level"></a>
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<h4>NLP Level</h4>
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<p>The Non-Linear Processing (NLP) setting determines the amount of non-linear
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suppression that will be applied in conjunction with the AEC algorithm
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for each channel. NLP will remove the residual echo not removed by the
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linear part of the AEC algorithm.</p>
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<p>This parameter represents a trade-off between achieving good double-talk
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performance, with no suppression of the local speech signal, and very
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robust echo suppression, with no echo audible on the far side. At its
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most aggressive setting (NLP at 100%), the non-linear processing will
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remove any of the residual far-side echo picked up by the microphone.
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However, this is done with an increased risk that some of the near-side
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speech will be degraded as well, especially during double-talk. At its
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least aggressive setting (NLP at 0%), the non-linear processing is effectively
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disabled, which may let some echo through, but will allow for a more natural
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double-talk performance.</p>
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<p>The best setting for this parameter may depend on several factors, including
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the acoustic properties of the room and user preference. The default value
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of 50% may give a good balance between these two competing goals.</p>
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<a name="NC_Level" id="NC_Level"></a>
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<h4>NC Level</h4>
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<p>The Noise-Cancellation (NC) setting will determine the amount of noise
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cancellation that will be applied to each channel. The noise cancellation
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algorithm is a very advanced algorithm that will remove steady-state noise
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without compromising the quality of speech passing through the channel.
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This algorithm is great for removing projector noise, HVAC, and other
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unwanted background noise that can compromise speech intelligibility.</p>
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<h4>Signal Threshold</h4>
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<p>In a conferencing system, some microphones may have a mute or push-to-talk
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feature built in. If a mic goes into or comes out of mute, then the characteristics
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of the conferencing system change instantly, and echo may leak through
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as the AEC re-converges. A signal threshold is defined to allow mics with
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mute or push-to-talk features to work seamlessly with AEC. Using the threshold,
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a level can be defined that is below the normal, ambient noise floor of
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the room. If the mic level goes below this level, then the AEC algorithm
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will treat the microphone as muted, and minimize any echo that would have
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occurred otherwise. The "Active" LED indicates that the microphone
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level is over the threshold, and the mic is not treated as being muted.
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When the LED is off, the mic level is below the threshold, and the mic
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will be treated as being muted.</p>
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<p>To set the threshold:</p>
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<ul>
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<li>Put the microphone in its muted mode.</li>
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<li>Set the threshold to a level where the LED turns itself on and
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off randomly.</li>
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<li>Raise the threshold from this level by 3 to 6 dB. The LED should
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now be off with no flickering.</li>
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<li>Take the microphone out of its muted mode and the LED should illuminate.</li>
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</ul>
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<p>This process may need to be repeated if the microphone's preamp gain
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is adjusted. To disable the mute feature based on signal threshold, simply
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set the threshold to its minimum value.</p>
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<a name="Advanced_Panel" id="Advanced_Panel"></a>
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<h4>Advanced Panel</h4>
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<p>The advanced panel gives access to controls for the automatic gain control.</p>
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<p> <img src="AEC_control_panel_2.JPG" alt="" width="223" height="654" border="0" class="hcp2"></p>
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<p><a name="AGC" id="AGC"></a></p>
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<h4>AGC</h4>
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<p>The Automatic Gain Control is designed for voice applications. It is
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designed to compensate for varying distances between the speaker and their
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microphone as well as speech level variances at the near end. This provides
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the far end with a signal that will automatically be increased or decreased
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to maintain a consistent audio level.   <span class="hcp1">Setting
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the maximum gain too high can cause inconsistent gain structures and bring
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up the noise floor</span>.  The AGC will adjust the gain during near
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side speech only. This means that during pauses in near side speech, the
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noise floor will maintain a constant level, and will not grow to hit a
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target gain output. Only near side speech signals are used to control
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the gain.</p>
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<p>To use the AGC, first define target levels for the transmitted speech
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signal. The default target levels for AGC are a maximum of 6dBu and a
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minimum of -10dBu, which define a target window with 16dB of dynamic range.
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If the speech level is within the target window already, then the AGC-applied
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gain will go to 0 dB.</p>
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<p>If the speech signal is below the target window (i.e., below the minimum
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target level), then the AGC will increase the gain (to a limit) so that
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the signal level meets the minimum target level. The AGC will limit the
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gain it can add to a signal by a maximum gain setting. Once the AGC has
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adjusted its gain high enough to meet the maximum gain setting, it will
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stop adding gain, even if the minimum target level is not reached. This
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is useful to stop very weak speech signals, such as whispering, from driving
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the gain too high.  <span class="hcp1">Setting the
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maximum gain too high can cause inconsistent gain structures and bring
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up the noise floor.</span></p>
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<p>Similarly, if the level of the speech signal is higher than the maximum
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target level, then the AGC will reduce the gain, by as much as the minimum
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gain setting, in an attempt to bring the speech level down to the maximum
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target level.</p>
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<p>A generous range for the maximum gain and the minimum gain have been
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provided. Care should be taken, particularly with the maximum gain setting,
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to avoid extreme levels. Situations where the maximum gain setting should
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be set over 10dB will be rare. The maximum gain setting has the potential
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to break a gain structure, so set it carefully, especially if the setting
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is to be used over 10dB.</p>
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<p>The attack and release rates for the AGC describe how fast it will adjust
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its gain. Because the AGC only adjusts gain during near-side speech signals
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(and not during unvoiced consonants like t, s, p, and f), the attack and
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release rates should be set higher than other typical AGC implementations.</p>
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<p>The AGC meter shows the current amount of gain being applied to the
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signal.</p>
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<p>Here is a summary of the AGC settings:</p>
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<ul>
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<li>AGC On  :                  On/Off
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switch to enable/disable automatic gain control</li>
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<li>AGC Max Target  :      Range -20
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dBu to +20 dBu (default is +6 dBu)</li>
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<li>AGC Min Target  :       Range
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-20 dBu to +20 dBu (default is -10 dBu)</li>
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<li>AGC Max Gain  :        Range
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0 dB to +20 dB (default is +6 dB)</li>
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<li>AGC Min Gain  :         Range
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-20 dB to 0 dB (default is -6 dB)</li>
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<li>AGC Max Rise Rate : Range 0.1 to 100 dB/s (default is 13.33
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dB/s)</li>
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<li>AGC Max Fall Rate :  Range 0.1 to 100 dB/s (default is
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15.62 dB/s)</li>
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</ul>
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<h2>AEC Card Properties</h2>
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<a name="Use_Meter" id="Use_Meter"></a>
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<h4>Use Meter</h4>
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<p>Will enable input channel metering for this card.</p>
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<h4>Enable Mutes</h4>
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<p>Se to YES will add input mutes to the control panel.</p>
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<a name="Metering_Point" id="Metering_Point"></a>
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<h4>Metering Point</h4>
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<p>Changes the point in the audio path used for metering. Can be pre or
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post AEC. The default is to meter pre-AEC so that the meter shows the
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level of audio being received by the AEC card before any processing is
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applied.</p>
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<h4>Full Bandwidth (20kHz)</h4>
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<p>You can select either one of the 2 versions of AEC algorithm to use.
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 Set to 'No' to select the original version AEC with limited bandwidth.
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 Set to 'Yes' to select the later version AEC algorithm with a higher
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bandwidth which will give an improved sound quality.</p>
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<p><span class="hcp1">Note:</span> Designs created in versions
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of London Architect before version 3.04 will default to 'No' for full
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bandwidth when loaded into London Architect.  Designs created from
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version 3.04 onward will default to 'Yes' for full bandwidth.  In
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a design which has more than one AEC card it is possible to have the bandwidth
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set individually for each card.</p>
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<a name="Parameters" id="Parameters"></a>
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<h4>Parameters</h4>
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<ul>
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<li>Reference - as per standard analogue input card</li>
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<li>Attack - as per standard analogue input card</li>
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<li>Release - as per standard analogue input card</li>
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<li>Gain - as per standard analogue input card</li>
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<li>Phantom Switch - as per standard analogue input card</li>
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<li>AEC Enable - Appears on the default panel, described in 'AEC Card
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Control Panel' section.</li>
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<li>NLP Enable - Non-linear processing is on by default. This parameter
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is for advanced use and is accessible via the design tree.</li>
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<li>NLP Level - Appears on the default panel, described in 'AEC Card
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Control Panel' section.</li>
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<li>NC Enable - Noise cancellation is on by default. This parameter
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is for advanced use and is accessible via the <a href="MAINDesign_tree.htm">design
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tree</a>.</li>
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<li>NC Level - Appears on the default panel, described in 'AEC Card
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Control Panel' section.</li>
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<li>Signal Threshold - Appears on the default panel, described in 'AEC
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Card Control Panel' section.</li>
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<li>AGC Enable - Appears on the default panel, described in 'AEC Card
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|
Control Panel' section.</li>
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<li>AGC Max Gain - This parameter sets the maximum gain for AGC operation.
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It is for advanced use and is accessible via the design tree.</li>
|
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|
<li>AGC Min Gain - This parameter sets the minimum gain for AGC operation.
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|
It is for advanced use and is accessible via the design tree.</li>
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|
<li>AGC Max Target - This parameter sets the maximum target for AGC
|
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|
operation. It is for advanced use and is accessible via the design
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|
tree.</li>
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|
<li>AGC Min Target - This parameter sets the minimum target for AGC
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|
operation. It is for advanced use and is accessible via the design
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|
tree.</li>
|
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|
<li>AGC Attack - This parameter sets the maximum rise rate (dB/s) for
|
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|
AGC operation. It is for advanced use and is accessible via the design
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|
tree.</li>
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|
<li>AGC Release - This parameter sets the maximum fall rate (dB/s)
|
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|
for AGC operation. It is for advanced use and is accessible via the
|
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|
design tree.</li>
|
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|
</ul>
|
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|
<h2>London Architect Configuration Symbol</h2>
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|
<p>For each recognized AEC card in a Soundweb London unit the following
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|
configuration symbols will appear in the Default Configuration view:</p>
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|
<p><img src="AEC_Input_Card.jpg" alt="" width="384" height="195" border="0" class="hcp2"></p>
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|
<p>The left hand 'AEC Input Card' block functions like a standard Soundweb
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|
London input card. This block contains the 4 channels of processed AEC
|
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|
audio as well as the 4 channels of 'dry' (unprocessed) input audio being
|
||
|
fed into the Soundweb London AEC card. The right hand block 'AEC Input
|
||
|
Ref (REFERENCE) Return' is used to provide the REFERENCE signal for each
|
||
|
AEC algorithm. The Reference signal is the <b>signal that will be removed
|
||
|
by the AEC algorithm from the signal path</b>. The Reference signal should
|
||
|
be taken from as close to the output as possible. This will provide the
|
||
|
AEC algorithm with the most accurate representation of the signal (to
|
||
|
be cancelled) and will provide the best AEC performance.</p>
|
||
|
<h4>Example: Basic Conferencing without Local Sound Reinforcement</h4>
|
||
|
<p>Local Sound Reinforcement refers to a design where the local microphones
|
||
|
feed both the far-side and the local room speakers. This typically applies
|
||
|
in large rooms where other participants located in the same room can't
|
||
|
hear the person speaking.</p>
|
||
|
<p>This example shows 4 microphones feeding the local audio to the far-side
|
||
|
via a <a href="Telephone_Hybrid_Cards.htm">Telephone Hybrid</a>. The far
|
||
|
side audio is received via the <a href="Telephone_Hybrid_Cards.htm">Telephone
|
||
|
Hybrid</a> and processed by the London's Low Pass, High Pass, and Parametric
|
||
|
EQ's before being sent to the local room's speaker(s) for the local participants
|
||
|
to hear. Once the far-side audio leaves the local-side's speakers the
|
||
|
signal will bounce around the room, re-enter the local microphone (mixing
|
||
|
with the local side's speech), and the far-side signal will be sent back
|
||
|
to the far-side resulting in 'echo'. To prevent this echo, the far-side
|
||
|
signal is sent to the Reference inputs of the AEC card where the AEC algorithm
|
||
|
will compare this signal with the input signals of the AEC card (the microphones)
|
||
|
and remove the Reference signal (far side signal) from the input signal
|
||
|
path resulting in only the local side audio being sent to the far-side
|
||
|
i.e. no echo.</p>
|
||
|
<p><img src="image15.jpg" alt="" width="800" height="263" border="0" class="hcp2"></p>
|
||
|
<p>The image below is the same design with the Reference signal wired incorrectly.
|
||
|
Because the Reference is taken before the room processing blocks, the
|
||
|
AEC algorithm will not understand that certain frequencies were cut/boosted
|
||
|
intentionally and will not be able to model the room to its full ability.</p>
|
||
|
<p><img src="image16.jpg" alt="" width="800" height="147" border="0" class="hcp2"></p>
|
||
|
<h4>Example: Basic Conferencing with Local Sound Reinforcement</h4>
|
||
|
<p>This example shows 4 microphones feeding the audio to the far side via
|
||
|
a 'Telephone Hybrid' as well as feeding the local speakers for local sound
|
||
|
reinforcement. Signal mixing is performed using the <a href="Gated_Automixer.htm">Gated
|
||
|
Automixer</a> processing object.  The best method for this type of
|
||
|
design incorporates a mix-minus setup to maintain proper gain structure,
|
||
|
and to prevent the speaker directly above the person talking from transmitting
|
||
|
a delayed - room-coloured - copy that will re-enter the open microphone
|
||
|
and be transmitted to the far side along with the original voice signal.</p>
|
||
|
<p>The design below shows the both the far side and near side signals feeding
|
||
|
the local room speakers. This design works, but as explained above, since
|
||
|
the Reference signal is not being fed after the room processing blocks
|
||
|
(as close to the speaker output as possible) the AEC algorithm won't perform
|
||
|
to its full potential.</p>
|
||
|
<p><img src="image20.jpg" alt="" width="800" height="284" border="0" class="hcp2"></p>
|
||
|
<p>If the Reference is moved to the same location as in the previous 'No
|
||
|
Reinforcement' example, it will satisfy the rule of placing the Reference
|
||
|
'as close as possible' to the speaker output, but in doing so the Reference
|
||
|
will be fed with a mix of both the near side and far side signals. Since
|
||
|
the Reference signal is the 'signal we want to remove from the input audio
|
||
|
path' then this means that the AEC algorithm will cancel the speaker's
|
||
|
voice coming into the AEC Input Card.  Since the input microphone
|
||
|
signal path is being fed to the far side as well as the local speakers
|
||
|
then the far side will not be able to hear the speaker either. (Typically
|
||
|
what happens is that only portions of the speaker's voice gets cancelled
|
||
|
because of the VAD state and it causes the voice to distort and sound
|
||
|
bad to both the far side and the near side)</p>
|
||
|
<p><img src="image21.jpg" alt="" width="800" height="314" border="0" class="hcp2"></p>
|
||
|
<p>To solve this dilemma utilize an N-input parametric EQ and another set
|
||
|
of High/Low Pass objects in order to provide the Reference the same signal
|
||
|
as the room speaker. This means that you only Reference (remove) the far
|
||
|
side signal while still feeding a mix of both near side and far side audio
|
||
|
to the room speakers. <span class="hcp1">It is very important
|
||
|
to make sure the same settings are maintained in both signal paths. In
|
||
|
particular, care must be taken that any non-linear processing (such as
|
||
|
compression or limiting) that happens to the speaker output signal, also
|
||
|
happens to the Reference signal</span>. BSS Audio recommend using the
|
||
|
'copy control values' feature to ensure the settings are identical.</p>
|
||
|
<p><img src="image22.jpg" alt="" width="800" height="301" border="0" class="hcp2"></p>
|
||
|
<h4>Example: Local Media Distribution</h4>
|
||
|
<p>Add to the previous Local Sound Reinforcement design by adding a local
|
||
|
DVD player and a local PC audio input for presentations that the near
|
||
|
side would like to share with the far side as well as having the near
|
||
|
side speakers distribute this material. This design requires that the
|
||
|
local media inputs (DVD/PC) are sent directly to the far side via a 'Telephone
|
||
|
Hybrid' to provide a high quality audio signal. For the near side to hear
|
||
|
the local media inputs through their speakers, Reference (remove) the
|
||
|
far side signal and local media signal from the microphone input signal
|
||
|
path to prevent the far side from hearing their own voice (echo) and from
|
||
|
getting a lower quality local media signal - which can be interpreted
|
||
|
as an echo as well. Remember that you are already directly sending the
|
||
|
local media signal to the far side so you need to make sure that the local
|
||
|
media signal is not allowed to travel to the far side via the microphone
|
||
|
input signal path as well.</p>
|
||
|
<p>Once again, it should be emphasized that the ERL meter needs to be in
|
||
|
the 'green' zone while the local Media sources are playing during the
|
||
|
conference or the AEC algorithm will not be able to remove the echoes.</p>
|
||
|
<p><img src="image23.jpg" alt="" width="1000" height="275" border="0" class="hcp2"></p>
|
||
|
<p> </p>
|
||
|
</body>
|
||
|
</html>
|