Voice Channel
Tube Channel Strip with Digital Connectivity

USERS GUIDE

IMPORTANT SAFETY INSTRUCTIONS READ FIRST
This symbol, wherever it appears, alerts you to the presence of uninsulated dangerous voltage inside the enclosure. Voltage that may be sufficient to constitute a risk of shock.
This symbol, wherever it appears, alerts you to important operating and maintenance instructions in the accompanying literature. Please read manual.
Read instructions: Retain these safety and operating instructions for future reference. Heed all warnings printed here and on the equipment. Follow the operating instructions printed in this user guide. Do not open: Aside from one vacuum tube, there are no user serviceable parts inside. Refer any service work to qualified technical personnel only. Power sources: Only connect the unit to mains power of the type marked on the rear panel. The power source must provide a good ground connection. Power cord: Use the power cord with sealed mains plug appropriate for your local mains supply as provided with the equipment. If the provided plug does not fit into your outlet consult your service agent. Route the power cord so that it is not likely to be walked on, stretched or pinched by items placed upon or against. Grounding: Do not defeat the grounding and polarization means of the power cord plug. Do not remove or tamper with the ground connection on the power cord. Ventilation: Do not obstruct the ventilation slots or position the unit where the air required for ventilation is impeded. If the unit is to be operated in a rack, case or other furniture, ensure that it is constructed to allow adequate ventilation. Moisture: To reduce the risk of fire or electrical shock do not expose the unit to rain, moisture or use in damp or wet conditions. Do not place a container of liquid on it, which may spill into any openings. Heat: Do not locate the unit in a place close to excessive heat or direct sunlight, as this could be a fire hazard. Locate the unit away from any equipment, which produces heat such as: power supplies, power amplifiers and heaters. Environment: Protect from excessive dirt, dust, heat, and vibration when operating and storing. Avoid tobacco ash, drink spillage and smoke, especially that associated with smoke machines. Handling: To prevent damage to the controls and cosmetics avoid rough handling and excessive vibration. Protect the controls from damage during transit. Use adequate padding if you need to ship the unit. To avoid injury to yourself or damage to the equipment take care when lifting, moving or carrying the unit. Servicing: Switch off the equipment and unplug the power cord immediately if it is exposed to moisture, spilled liquid, objects fallen into opening, or the power cord or plug becomes damaged during a lightning storm or if smoke odor or noise is noted. Refer servicing to qualified technical personnel only. Installation: Install the unit in accordance with the instructions printed in the user guide. 1
IMPORTANT SAFETY INSTRUCTIONS READ FIRST... 1 INTRODUCTION.... 4 INSTALLATION..... 4
AC Power Hookup...... 4 Analog Audio Connections..... 4
FRONT PANEL CONTROLS and JACKS... 5
Mic Preamp Controls...... 5 Instrument Input...... 5 Gain Control...... 5 Impedance Control...... 5 Pad Switch...... 5 Phantom Power...... 5 Invert Switch...... 5 Low Cut Switch...... 5 Tube Voltage Switch...... 6 Dynamic Processor Controls..... 7 Threshold Control...... 7 Ratio Control...... 7 De-esser Control..... 7 De-esser Freq. Control...... 7 Attack Control....... 7 Release Control...... 8 Gain Reduction LED Meter..... 8 Expander/Gate Switch...... 8 Expander/Gate Threshold Control...... 8 Semi-Parametric EQ...... 9 EQ Position Switch...... 9 EQ Bypass Switch...... 9 Output Level Control...... 10 Output Level Meters...... 10 Sample Rate/Dither Control..... 11 Optical Output Switch...... 11

REAR PANEL CONNECTIONS.... 12
Mic/Line Input...... 12 Mic Preamp Output...... 12 Insert Input...... 12 Balanced Output...... 12 A/D Main Insert and A/D CH2 Insert...... 12 Wordclock Input and Thru Jacks...... 12 ADAT Input Jack...... 13 Optical Output Jack...... 13 S/PDIF Output Jack...... 13 AES/EBU Output Jack..... 13 USB Jack....... 13

APPLICATIONS..... 14

Bypassing Components Of The Voice Channel..... 14 Optimizing The Preamp For Lowest Noise.... 14 Utilizing Pre/Post Compression EQ..... 14
WARRANTY INFORMATION.... 15 SERVICE..... 16 SPECIFICATIONS..... 17
FIGURE FIGURE FIGURE FIGURE FIGURE FIGURE FIGURE

LIST OF FIGURES

Preamp section..... 6 Dynamics Section.... 6 Equalizer Section.... 9 Signal Flow Block Diagram... 10 Output Section.... 11 Rear Panel.... 11 Wordclock Termination.... 13

INTRODUCTION

The ART Voice Channel is the answer to your recording and computer audio interface needs. Our secondgeneration discrete Class-A microphone preamp provides clean quiet gain while maintaining incredible transparency. A powerful dynamics processor subtly controls transients and noise of the most demanding sources. The ART Voice Channels semi-parametric EQ offers wide tune-ability and can be patched before OR after the dynamics processor. Separate insertion jacks allow you to use your favorite external signal processing gear immediately after the Mic preamp and before the EQ and dynamics processor. Another patch point exists just before the A/D converters. Choose between a wide range of outputs including balanced analog output, 44.1 KHz to 192 KHz AES/EBU, S/PDIF, ADAT and USB. Both analog and digital meters provide a detailed indication of audio levels.

INSTALLATION

The ART Voice Channel may be used in a wide variety of applications and environments. In a rackmountable, all-steel enclosure, the unit is designed for continuous professional use. Mounting location is not critical, however for greater performance reliability we recommend that you not place the unit on top of power amps, or other sources of heat and/or strong magnetic fields. The tube circuitry needs about a minute to warm up and stabilize from a cold power up.

AC Power Hookup

The ART Voice Channel has an internal power supply. Only connect the unit to mains power of the type marked on the rear panel. The power source must provide a good ground connection, and the ground pin on the mains plug should never be defeated.

Analog Audio Connections

Audio connections to and from the Voice Channel are: Front panel balanced combo input: [XLR] Pin 2 = Hot (+), Pin 3 = Cold (-), Pin 1 = Ground [1/4] Tip = Hot (+), Sleeve = Ground Rear panel balanced combo input: [XLR] Pin 2 = Hot (+), Pin 3 = Cold (-), Pin 1 = Ground [1/4] Tip = Hot (+), Ring = Cold (-), Sleeve = Ground Rear panel balanced 1/4 output: Rear panel 1/4 insert input: Mic Preamp Output: A/D Main & A/D CH2 Inserts: Tip = Hot (+), Ring = Cold (-), Sleeve = Ground Tip = Hot (+), Sleeve = Ground Tip = Hot (+), Sleeve = Ground Tip = Input, Ring = Output, Sleeve = Ground
FRONT PANEL CONTROLS and JACKS

Mic Preamp Controls

The Voice Channel input consists of a discrete Class-A differential preamp. The circuit is optimized for low impedance microphones as well as line level signals. Up to 60dB of gain is available from this stage. The output can be inverted using the INVERT switch. The impedance of the front and rear XLR inputs is continuously variable for fine-tuning the preamp to a wide variety of mics. Phantom power is available on the XLR inputs as well. A selectable low-cut filter removes rumble, wind noise, and pops, thereby increasing clarity.

Instrument Input

The 1/4 T/S jack on the front panel provides a high impedance unbalanced input, and when used, automatically switches off the mic pre-amp. (The rear combo jacks 1/4 T/R/S balanced input is lower impedance and is part of the mic pre-amp. The rear jack is not intended to be used with high impedance microphones or instruments.) NOTE: When using the INSTRUMENT INPUT, the PAD switch is disabled and does NOT affect the gain.

Gain Control

This control adjusts both the mic pre-amp gain as well as the instrument input gain. The gain marked applies to the mic pre-amp without the PAD switch depressed. The instrument input gain markings are on the right side of the slash (/). Refer to the APPLICATIONS section to learn how to optimize the gain control for low noise operation.

Impedance Control

This knob sets the load impedance at the front and rear panel XLR inputs of the Voice Channel. Use the IMPEDANCE CONTROL to subtly tune the sound of your microphone. Various microphones will change their sound at differing load impedances. The correct setting is subjective. Adjust this control to personal taste.

Pad Switch

This switch reduces the mic pre-amp gain by up to 20dB to prevent clipping when high level mic, or line level signals are applied to the balanced XLR or 1/4 T/R/S inputs. This switch does NOT affect the 1/4 T/S front panel INSTRUMENT INPUT.

Phantom Power

The switch safely applies +48Volt phantom power to the XLR inputs. Use phantom power only when the microphone that you are using requires it. Doing so will extend the life of the Voice Channel as well as reducing the possibility of shock hazard.

Invert Switch

This switch selects the output phase of the Voice Channel. There is a 180 degree phase shift through the Voice Channel when lit.

Low Cut Switch

This switch inserts a 100Hz 6dB/Oct. Low-Cut filter into the signal path. The filter is designed to remove rumble, pops, and wind noise, yet still sound natural.

FIGURE 1 Preamp section

Tube Voltage Switch
The vacuum tube preamp section can be adjusted to run at two different plate voltages. Refer to Figure 2 for the location of the switch. Choose the NORMAL setting for adding warmth to the input signal. This setting has an increased amount of tube saturation at higher signal levels. Choose the HIGH setting to increase overall gain, headroom, and bandwidth. NOTE: The change between tube voltage modes is gradual, taking 10 to 20 seconds to be fully activated.
FIGURE 2 Dynamics Section
Dynamic Processor Controls
The ART Voice Channel dynamics section consists of an above threshold Compressor/Limiter with Deesser plus a selectable Expander/Gate. The attack and release controls allow a wide range of adjustment while the complex detector assures fast response without distortion. The De-esser is frequency tunable.

Threshold Control

This control sets the level, above which the Compressor/Limiter in the Voice Channel starts to act on the input signal. As the control is turned clockwise, more input signal is required to begin reducing gain. The compression action can be seen in the Gain Reduction LED meter.

Ratio Control

The RATIO control sets the amount of gain reduction that takes place based on how far the input signal is over the threshold level (set by the THRESHOLD control). When the control is fully counterclockwise, the Compressor/Limiter is OFF. A good starting point for vocals is 2.5:1. To have the unit act as a limiter, set the RATIO control to 20:1.

De-esser Control

The DE-ESSER control sets how much more the gain is reduced at high frequencies when using the Compressor/Limiter. The most common application is reducing sibilance when compressing vocals. When fully counterclockwise, the De-esser function is OFF. As the control is turned clockwise, high frequency material is compressed more than mid and low frequency material.

De-esser Freq. Control

This control selects which high frequencies the DE-ESSER acts upon. Turned fully counterclockwise, the DEESSER acts on the upper mid-range. When set fully clockwise, only the highest frequencies are reduced more during de-essing compression. Center the DE-ESSER FREQ. control as a starting point for vocal work.

Attack Control

The ATTACK control sets the time it takes the Compressor/Limiter to respond to increases in signal level (by reducing gain). You can use this control to shape the front end of the dynamics envelope. One example is to listen to a snare hit and adjust the attack control. A short attack makes the snare sound thin. As the attacks go longer (and the knob is turned clockwise) you should hear more of the thump in the compressed snare. The downside is that this creates an overshoot, (a large transient), the length of which is the time set by the ATTACK control. Overshoots less than 1 msec are very hard to hear even when they are clipped. If the attack is set too fast, the gain may be reduced too much and thereby create a pumping sound1. One way to eliminate this is to use the LOW CUT filter to remove plosive sounds in vocals that can make the detector overreact.
Pumping in a Compressor/Limiter sounds like the signal is muted when it shouldnt be. 7

Release Control

The RELEASE control sets the time the Compressor/Limiter takes to increase the gain after the input level drops. Longer settings maintain the dynamics of the input signal, while shorter settings reduce the dynamics. Shorter settings will also increase the apparent reverberation, and at extreme gain reduction settings, lead to breathing artifacts2

Gain Reduction LED Meter

The GAIN REDUCTION meter displays the Compressor/Limiters attenuation action. The meter covers a very large range while offering high resolution. The large yellow LED at the right-hand end of the meter indicates Expander or Gate action. The brightness of the LED indicates the amount of gain reduction for the Expander function. Since the Gate is either ON or OFF, there is no brightness variation.

Expander/Gate Switch

This switch allows the selection of the Expander or the Gate functions. Both are useful in reducing unwanted background noise in the audio signal. In the OUT position the Expander function is selected. Use this mode to gradually reduce background noise and maintain some of the input dynamics. This is useful for instruments with gradually decaying amplitude envelopes. Depress this switch to select the Gate mode. This mode quickly cuts off the noise as the input signal drops.
Expander/Gate Threshold Control
The Expander/Gate action begins below the level indicated by the EXPANDER/GATE THRESHOLD control. The EXPANDER/GATE THRESH. LED in the GAIN REDUCTION display will light when the Expander or Gate attenuates the input signal. The Expander/Gate Threshold detector has built-in hysteresis, which causes the unit to trigger ON at a higher level than the level required to trigger back to the OFF state. The Expander slope is about 1:1.5. This is subtle enough to maintain the decay envelope of the source material and still lower the noise as the input signal drops. The EXPANDER/GATE THRESH. LED will light dimly for the first 5dB of gain reduction, and then glow brightly as the attenuation increases above this level. The Gate function has an intelligent detector with a fast attack and release, coupled with a program dependant hold. The hold time is longer for sustained passages and shorter for transients. As the input drops below the threshold and the input signal is muted, the EXPANDER/GATE THRESH. LED lights brightly. You can turn off the Expander/Gate function by setting the THRESHOLD control fully counterclockwise to OFF.
Breathing is the sound of the Compressor/Limiter turning up the gain so quickly you can hear breathing noises between words during vocal processing. 8

Semi-Parametric EQ

The ART Voice Channel offers a four-band semi-parametric equalizer. The EQ can be bypassed as well as positioned before or after the dynamics processing section. Each band has + 15dB of control range. The High and Low EQ bands are shelving type with a switch selectable turnover point. The two Mid bands can be continuously tuned over a five octave range.
FIGURE 3 Equalizer Section

EQ Position Switch

The EQ POSITION switch allows you to connect the EQ before or after the dynamics processing block. This is useful in cases where the input signal needs a great deal of EQ before the Compressor/Limiter processes it. One example of using the EQ in the PRE position is using the LOW EQ control as a tunable low frequency cut, supplementing the LOW CUT filter switch. Refer to Figure 4 for the block diagram of the Tube Channel. Note that the Mic preamp insert jack function is located before the EQ position switch.

EQ Bypass Switch

This switch allows you to instantly set the EQ completely flat without loosing the current EQ settings.
FIGURE 4 Signal Flow Block Diagram

Output Level Control

The OUTPUT LEVEL control provides gain or attenuation to adjust for a variety of system operating levels. This control affects the levels sent to the A/D converter and to the balanced analog OUTPUT jacks.

Output Level Meters

The ART Voice Channel provides both analog and digital output meters. The meters monitor the signal level just after the output control. This signal is sent to both the analog and digital outputs. 0 VU on the analog VU meter corresponds to +4dBu on the balanced outputs, and about 20dB on the LED bargraph meter. The LED bargraph meter indicates peak levels as well as average levels. Average levels are indicated by a continuous string of LEDs being lit. Peak levels are indicated by a single LED and are held for about 2 seconds. The last LED in the meter is marked Clip, and it indicates that the output level is set too high. The A/D Clip indicator LEDs act independent of the OUTPUT LEVEL meter. This provides an accurate indication of A/D converter clipping. This is useful when using the A/D insertion jacks, since the level at the converters will not be indicated on the main meters when these inputs are used.

FIGURE 5 Output Section

Sample Rate/Dither Control
The SAMPLE RATE/DITHER knob selects the sample rate for the AES/EBU, S/PDIF, and optical outputs. It also selects the dither applied. Set the switch appropriately to match up with 16 or 24 bit encoding. NOTE: This control does NOT affect the USB output.

Optical Output Switch

The OPTICAL OUTPUT switch sets the signal format of the rear panel OPTICAL OUTPUT connector. In ADAT mode channels 1 and 2 are the left and right A/D outputs of the Voice Channel respectively. NOTE: If the A/D insertion jacks are not being used, both channels 1 and 2 carry the same signal. If the ADAT INPUT is also being used, channels 3 thru 8 are passed through along with channels 1 and 2 of the ART Voice Channel.

FIGURE 6 Rear Panel

REAR PANEL CONNECTIONS

Mic/Line Input

This combo jack provides balanced inputs to the microphone preamplifier. The XLR connection is in parallel with the front input jack XLR. The input impedance of both XLR connections is variable by the front panel Impedance knob. The rear 1/4 input of the combo jack overrides the front panel XLR input when used. This inputs impedance is NOT affected by the front panel Impedance control and is fixed at 20K Ohms. The Front panel Instrument input overrides the rear jack when it is used.

Mic Preamp Output

This 1/4 T/S unbalanced jack provides a direct signal from the microphone preamplifier, ahead of the EQ and dynamics processors. This output can be used in conjunction with the INSERT INPUT jack to insert external signal processors between the main preamp section and the EQ and dynamics processing of the Voice Channel.

Insert Input

This 1/4 T/S unbalanced jack is an input to the EQ and dynamics processing sections. This input can be used in conjunction with the MIC PREAMP OUTPUT jack to insert external signal processors between the main preamp section and the EQ and dynamics processing of the Voice Channel.

Balanced Output

The analog output of the Voice Channel is available on both a 1/4 TRS balanced jack and an XLR jack. This output is active balanced, and will adjust to balanced or unbalanced termination without gain change. The LED and analog meter monitor the level present at this output. 0 VU on the analog meter corresponds with +4dBu (about 1.2 Volts RMS).
A/D Main Insert and A/D CH2 Insert
Signal processing can be added between the analog output of the Voice Channel and the left and right channels of the A/D converter by using the A/D MAIN INSERT and the A/D CH2 INSERT respectively. Use a 1/4 T/R/S (stereo) cable. The Ring is the output of the preamp and the Tip is the input to the A/D converter. In order to use just the A/D converter and not the Voice Channel preamp, simply plug a standard 1/4 T/S phone cable into either A/D insertion jack.
Wordclock Input and Thru Jacks
The WORDCLOCK INPUT is used to externally sync the Voice Channel to a master clock source. The BNC WORDCLOCK INPUT jack is connected directly to the BNC WORDCLOCK THRU jack, providing the ability to loop through the Voice Channel and connect other devices to the wordclock sync source, saving the use of a BNC Tadapter. The input is high impedance thus leaving the wordclock connection unterminated. (A 75 Ohm BNC terminator should be used on the WORDCLOCK THRU jack if the WORDCLOCK INPUT jack is used only by itself.) Select the EXT/16 or EXT/24 sample rate setting on the front panel to utilize External Wordclock mode. Refer to FIGURE 7 for wordclock termination examples.

FIGURE 7 Wordclock Termination

ADAT Input Jack

The optical ADAT input allows the Voice Channel A/D converter to synchronize to systems using ADAT optical connections. The Voice Channel inserts its output in channels 1 and 2 of the ADAT stream while passing through channels 3 thru 8. Select ADAT/16 or ADAT/24 with the Sample Rate control on the front panel to enable this mode.

Optical Output Jack

The OPTICAL OUTPUT jack works in conjunction with the front panel OPTICAL OUTPUT switch, to output either an ADAT formatted signal or a TOS formatted signal. The front panel SAMPLE RATE/DITHER control sets the sample rate, dither, and sync source for this output.

S/PDIF Output Jack

This connector provides S/PDIF formatted digital outputs from the left and right A/D converters. The front panel SAMPLE RATE/DITHER control sets the sample rate, dither, and sync source for this output.

AES/EBU Output Jack

This connector provides AES/EBU signal level digital outputs from the left and right A/D converters. The front panel SAMPLE RATE/DITHER control sets the sample rate, dither, and sync source for this output.

USB Jack

The USB jack provides the output of the Voice Channel to a direct computer USB connection. The Voice Channel will be recognized as a standard audio device on the PC or Mac. The sample rate and bit depth of this interface is set by the computer and is independent of the front panel settings. The audio data formats are limited to 32 KHz, 44.1 KHz, 48 KHz, 16 or 24 bit encoding.

APPLICATIONS

Bypassing Components Of The Voice Channel
To bypass the vacuum tube microphone preamp: Use the preamp INSERT INPUT jack. To bypass the Compressor/Limiter: Set the RATIO control fully counterclockwise to 1:1. To bypass the Expander/Gate: Set the EXPANDER/GATE THRESHOLD control fully counterclockwise to OFF. To bypass the EQ: Use the EQ bypass switch.
Optimizing The Preamp For Lowest Noise
The preamp of the ART Voice Channel can be optimized for low noise by combining use of the PAD and Input GAIN control for mic and line level signals. NOTE: The PAD control has no effect on the INSTRUMENT INPUT (Front panel 1/4 input of the INPUT combo jack). First, bypass the Compressor/Limiter, Expander/Gate and EQ. Next center the OUTPUT LEVEL control to 0 dB of gain. The OUTPUT LEVEL LED meter can now be used to correctly indicate the clip level of the input stage. Second, start with the PAD in the OUT position and the GAIN control centered. Refer to the OUTPUT LEVEL LED meters peak-hold function. Make sure that this meter never indicates clipping (the red LED is held on after a transient). The peak-hold indicator can be in the yellow range or in the 5dB range of levels. If the signal level is too high, depress the PAD switch. Third, adjust the GAIN control to keep the peak levels in the 5dB range of the OUTPUT LEVEL LED meter.

Utilizing Pre/Post Compression EQ
The Equalizer section of the Voice Channel can be applied either before OR after the Compressor/Limiter. This function is useful in getting the best performance out of the unit. Setting the Equalizer to PRE COMP is useful when the input signal contains too much low or high frequency information. Compressors in general work best when the audio is equalized first. (This can also serve to better control signal overshoots to the A/D converters as well.) One example is a vocal where the performer/microphone combination produces a popping sound, and when compressed, the Compressor/Limiter pumps. Sometimes the use of the LOW CUT filter in the preamp section does not cut enough of this out, or cuts too much of the lower midrange out of the signal to be useful. Here, the EQ can surgically remove this information and better optimize the overall sound.

WARRANTY INFORMATION

Limited Warranty:
Applied Research and Technology will provide warranty and service for this unit in accordance with the following warrants: Applied Research and Technology, (A R T) warrants to the original purchaser that this product and the components thereof will be free from defects in workmanship and materials for a period of three years from the date of purchase. Applied Research and Technology will, without charge, repair or replace, at its option, defective product or component parts upon prepaid delivery to the factory service department or authorized service center, accompanied by proof of purchase date in the form of a valid sales receipt.

Exclusions:

This warranty does not apply in the event of misuse or abuse of the product or as a result of unauthorized alterations or repairs. This warranty is void if the serial number is altered, defaced, or removed. A R T reserves the right to make changes in design or make additions to or improvements upon this product without any obligation to install the same on products previously manufactured. A R T shall not be liable for any consequential damages, including without limitation damages resulting from loss of use. Some states do not allow limitations of incidental or consequential damages, so the above limitation or exclusion may not apply to you. This warranty gives you specific rights and you may have other rights, which vary from state to state. For units purchased outside the United States, an authorized distributor of Applied Research and Technology will provide service.

SERVICE

The following information is provided in the unlikely event that your unit requires service. 1) Be sure that the unit is the cause of the problem. Check to make sure the unit has the proper power supplied, all cables are connected correctly, and the cables themselves are in working condition. 2) If you find the unit to be at fault, write down a complete description of the problem, including how and when the problem occurs. 3) Contact our Customer Service Department at (716) 297-2920 for your Return Authorization number or questions regarding technical assistance or repairs. Customer Service hours are 9:00 AM to 5:00 PM Eastern Time, Monday through Friday. 4) Pack the unit in its original carton or a reasonable substitute. The packing box is not recommended as a shipping carton. Put the packaged unit in another box for shipping. Print the RA number clearly on the outside of the shipping box. Print your return shipping address on the outside of the box. 5) Include with your unit: a return shipping address (we cannot ship to a P.O. Box), a copy of your purchase receipt, a daytime phone number, and a description of the problem. 6) Ship your unit (keep your manual!) to: Yorkville Sound 4625 Witmer Industrial Estates Niagara Falls New York 14305 ATTEN: REPAIR DEPARTMENT RA# ____________________

Fill in the following information for your reference: Date of purchase ___________________ Purchased from ___________________
Serial number __________________

SPECIFICATIONS

Input Impedance
Mic.... 150 to 3.4K Ohms, variable Line.... 20K Ohms Instrument.... 2.5M Ohms Preamp Insert... 7.5K Ohms A/D Inserts.... 10K Ohms

Output Impedance

Balanced Outputs... 200 Ohms balanced Preamp Output... 100 Ohms A/D Inserts.... 510 Ohms

Frequency Response

Analog In to Analog Out... 12 Hz to 100 KHz +0, -1 dB Analog In to Digital Out... 12 Hz to 20 KHz +0, -1 dB @ 44.1 KHz sample rate.... 16 Hz to 42 KHz +0, -1 dB @ 96 KHz sample rate
1 KHz.... <.015% typical 20 to 20 KHz... <.033% typical

Equivalent Input Noise

Mic/Line.... -130 dBu, Input shorted, Max gain, A weighted Instrument.... -105 dBu, Input shorted, Max gain, A weighted

Maximum Input Level

Mic/Line.... +18 dBu balanced with PAD Instrument.... +15 dBu

Maximum Gain

Mic.... 70 dB (XLR to balanced output) Instrument.... 40 dB (1/4 to balanced output)

Maximum Output level

Balanced.... +20 dBu Unbalanced... +20 dBu Output Level At Meter 0 VU.. +4 dBu

Preamp

Microphone Gain... 0 dB to +60 dB Instrument Gain.... +3 dB to +40 dB Low Cut Filter... 100 Hz, 1-pole, 6 dB/Octave
Boost/Cut... +12 dB on each band Low Freq. Tuning... 50/150 Hz Selectable MID 1 Freq. Tuning... 100 Hz to 3 KHz continuously variable MID 2 Freq. Tuning... 500 Hz to 15 KHz continuously variable High Freq. Tuning... 5K/15 KHz Selectable

Compressor/Limiter

Attack Time... 250 uSec. to 100 mSec. Release Time... 100 mSec to 3 Sec. De-esser Tuning... 2.5 KHz to 15 KHz continuously variable Compression Ratio.... 1:1 to 20:1 Expander Slope.... 1:1.5

Digital section

Wordclock Range... 30 KHz to 204 KHz Sample Rates... 44.1 KHz, 48 KHz, 88.2 KHz, 96 KHz, 176 KHz, 192 KHz A/D Dynamic Range... 106 dB A weighted USB A/D Dynamic Range... 94 dB A weighted
Dimensions... 3.50 H x 19.0 W x 9.17 D Weight...10.5 lbs. Power Requirements... USA 105 to 125 VAC/ 60 Hz Export units configured for country of destination.

USB Minimum System Requirements.. USB 1.1 compliant, Windows 98SE or newer, Mac OS 9.1 or newer
Note: 0 dBu = 0.775 VRMS, 0 dBV = 1 VRMS ART maintains a policy of constant product improvement. ART reserves the right to make changes in design, or make additions to, or improvements upon, this product without any obligation to install same on products previously manufactured. Therefore, specifications are subject to change without notice.
www.artproaudio.com E-mail: support@artproaudio.com
2006 Applied Research & Technology 165-5004-103

*Recent

in-mine EIM propagation measurements made by C o l l i n s do show an
unexpected propagation enhancement f o r frequencies near 500 k i l o h e r t z.
It is n o t known how c o n s i s t e n t t h i s enhancement w i l l be over a range
of c o a l mine environments, b u t should i t be found t o b e c o n s i s t e n t , then t h e problem of achieving adequate s i d e - l i n k communication range
w i l l be considerably a l l e v i a t e d.
However, i t appears t h a t t h e mech-
anism t h a t provides t h i s enhancement of s i d e - l i n k propagation w i l l s e r v e t o make v e r t i c a l propagation more d i f f i c u l t and thus t h e prob-
l e m of subsurface t o s u r f a c e communication may become worse under

such conditions.

Arthur D Little, lnc
One of the means proposed for aiding in overcoming these severe power and range limitations has been the application of voice bandwidth compression techniques. We anticipated that such techniques would permit greater range with the same transmitter power or the same range with less transmitter power. The purpose of this study was to determine the applicability of state-of-the-art voice bandwidth compression techniques for mine wireless voice communication systems as one means of reducing the power requirements of portable units. One reason why voice bandwidth compression appeared attractive is that at VLF to MF frequencies the noise received by a portable receiver in a mine is largely dominated by external mine-generated noise and not by receiver-generated noise. Therefore, if the received noise power could be lessened by reducing the bandwidth required for the voice signal, without paying a corresponding penalty in processing power and/or required signal-to-noise ratio, the desired range could be either extended with the same transmitter power or obtained with less power. Hence, a 10:l reduction in receiver-bandwidth could be expected to yield a 10:l reduction in received noise and hence a 10:l improvement in signal-to-noise ratio for the same amount of transmitter power. Extending this simplistic notion even further, a 100:l bandwidth reduction could be expected to yield a 100:l reduction in transmitter power. These potential values of transmitter power reduction were large enough to undertake this assessment of the potentials of state-of-the-art voice bandwidth compression techniques for achieving the goal of significantly reduced transmitter power consumption for mine wireless communication applications.

t a l systems can encode t h e analyzed v e r s i o n of t h e speech i n t o 1200 bps

Arthur D Littlelnc

and, i n some extreme c a s e s , i n t o as l i t t l e a s 700 bps, compression of 40:l i s o b t a i n a b l e by t h e s e systems.

This i n d i c a t e s a

However, t h e more
advanced systems of t h i s type r e q u i r e considerable computer power, p l a c i n g them considerably beyond t h e i r a p p l i c a t i o n t o mine communication requirements,
Techniques voice bandwidth compression techniques,
I n the a r e a of non-real-time
we f e e l t h e r e may be some m e r i t t o t h e stretched-out v o i c e system wherein an o r i g i n a l message T seconds long is i n e f f e c t s t r e t c h e d t o NT seconds long f o r transmission,
A l l t h e q u a l i t y of t h e o r i g i n a l speech may be
preserved a t t h e same signal-to-noise
r a t i o a s would be required f o r Thus, t r u e t r a n s m i t t e r
f u l l bandwidth transmission, b u t w i t h a bandwidth t h a t i s shrunk i n d i r e c t proportion t o t h e s t r e t c h - o u t f a c t o r. power requirement savings can be made, but only a t t h e expense of a delay between t h e t i m e. t h e message is spoken and t h e t i m e i t i s a v a i l a b l e t o be heard a t t h e r e c e i v e r ,
CONCLUSIONS From our study we conclude t h a t real-time state-of-the-art voice How-
bandwidth compression techniques a r e not s u i t a b l e a s a p r a c t i c a l means f o r improving mine wireless communications performance a t p r e s e n t. ever, non-real-time v o i c e bandwidth compression techniques and real-time, W conclude t h a t e
non-voice systems do o f f e r means f o r meeting t h e needs of through-thee a r t h emergency subsurface t o s u r f a c e communications. signal-to-noise systems.
t h e advantage of l i m i t i n g t h e voice s i g n a l waveform t o improve received r a t i o should be applied t o mine w i r e l e s s communication
R C M E D T IONS E O MN A Based on t h e f i n d i n g s and conclusions of our study, we recommend
t h a t t h e stretched-out voice systems be f u r t h e r explored as a means of meeting c e r t a i n of t h e mine communication needs, p a r t i c u l a r l y f o r l i n k s through t h e e a r t h f o r emergency purposes, and most important, t h e l i n k from underground t o t h e surface.
It i s p a r t i c u l a r l y t r u e t h a t power
savings h e r e can be important because of t h e l i m i t e d amount of energy a v a i l a b l e t o a miner trapped underground and t h e requirements f o r i n t r i n s i c s a f e t y f o r e l e c t r i c a l communication equipment,

H i s s i n g sounds t h a t do n o t go through t h e
f i l t e r ( s s s s and s h ) a r e n o i s e - l i k e , w i t h a v e r y wide frequency band. Voiced sounds look l i k e a t r a i n of impulses w i t h a fundamental requency between 60 and 240 Hz ( f o r men, b u t h i g h e r f o r women and c h i l d r e n ) w i t h a r i c h harmonic s t r u c t u r e t h a t f a l l s o f f w i t h frequency up t o perhaps 4000 Hz. These harmonics a r e modified by t h e f i l t e r and t h e formants
form a n envelope on t h e harmonics. Formants d i f f e r from i n d i v i d u a l t o i n d i v i d u a l , making t h e v o i c e s of different individuals distinctive. However, the combined e f f e c t s o f Figure
harmonics and f i l t e r always produce a r a p i d l y f a l l i n g spectrum.
1 shows t h e envelope of s p e c t r a f o r male s p e a k e r s and i l l u s t r a t e s t h i s
feature. One of t h e most s t r a i g h t f o r w a r d ways of d e c r e a s i n g t h e frequency r e q u i r e d f o r v o i c e t r a n s m i s s i o n i s simply t o chop some of i t o f f. s i v e t e s t s have been conducted on t h e i n t e l l i g i b i l i t y of t e l e p h o n e speech a s a f u n c t i o n of bandwidths. Exten-

Arthur D.Little,Inc.

Frequency in Hertz

Source: Ref. 3

FIGURE 1
ENERGY DENSITY SPECTRUM FOR MALE SPEECH

Arthur D Little, Inc.

F i g u r e 2 shows t h e e f f e c t o f changing t h e upper end o f t h e speech band w i t h a v a r i a b l e c u t o f f low-pass f i l t e r. Xf t h e band i s r e s t r i c t e d t o 1000 Hz, i n t e l l i g i b i l i t y i s d e c r e a s e d t o % , b u t 83% o f t h e energy

is still present.

T h i s c u r v e shows t h a t low f r e q u e n c i e s c o n t r i b u t e t o F i g u r e 3 shows t h e e f f e c t o f filter. While i n -
energy much more t h a n t o a r t i c u l a t i o n.
e l i m i n a t i n g low f r e q u e n c i e s w i t h a v a r i a b l e c u t o f f high-pass
T h i s shows t h a t e l i m i n a t i o n of f r e q u e n c i e s below 900 Hz l e a v e s i n t e l l i g i b i l i t y above 90%, b u t d e c r e a s e s energy t o 17% of t h e t o t a l. t e l l i g i b i l i t y is preserved, speaker r e c o g n i z a b i l i t y d e t e r i o r a t e s severely.

Arthur D Little,Inc.

Filter Cutoff Frequency, Hz
Source: Ref. 4 FIGURE 2 ENERGY AND ARTICULATION FOR SPEECH SIGNALS WITH LOW-PASS Fl LTER

Articulation

Source: Ref. 4

FIGURE 3

ENERGY AND ARTICULATION FOR SPEECH SIGNALS WITH HIGH-PASS FILTER
(This page i n t e n t i o n a l l y l e f t blank.)

Arthur D Little lnc.

METHODS OF VOICE BANDWIDTH COMPmSSION 1.

s e r v e d i s t h e z e r o c r o s s i n g s of t h e o r i g i n a l speech. o b l i t e r a t e d and t h e q u a l i t y i s u n p l e a s a n t.
l i t t l e i n t e l l i g i b i l i t y i s l o s t , b u t s p e a k e r r e c o g n i z a b i l i t y i s almost
An improvement i n e f f e c t i v e s i g n a l power o f a s much a s 12 dB can b e
o b t a i n e d from c l i p p i n g.
T h i s r e p r e s e n t s a l a r g e number of w a t t s a t t h e
power l e v e l s we a r e c o n s i d e r i n g.
Frequency D i v i s i o n (Stretched-out Voice) One of t h e most e f f e c t i v e ways of d e c r e a s i n g bandwidth i s t o d i v i d e
a l l f r e q u e n c i e s by a c o n s t a n t b e f o r e t r a n s m i s s i o n and m u l t i p l y them back t o t h e i r o r i g i n a l values a t the receiver. c o n s t a n t used. I f a l l c y c l e s of a l l frequenc i e s a r e t r a n s m i t t e d , t h i s i m p l i e s a slowing down of t r a n s m i s s i o n by t h e An elementary way t o a c h i e v e t h i s i s t o r e c o r d speech on
A t t h e r e c e i v e r t h e speech i s a g a i n
a t a p e r e c o r d e r and p l a y i t o v e r t h e l i n e a t a slower speed w i t h a c o r r e sponding d e c r e a s e i n bandwidths.
recorded and played back s u f f i c i e n t l y speeded up t o r e c o n s t i t u t e t h e original.
In p r i n c i p l e , t h e r e i s no l i m i t t o t h e s a v i n g s i n bandwidth a c h i e v a b l e
i n t h i s way. From an information theory viewpoint t h e r e i s a b s o l u t e l y no Time and bandF u r t h e r p r o c e s s i n g may reduce t h e a c t u a l channel s a v i n g i n channel c a p a c i t y s i n c e a g r e a t e r time i s used. w i d t h s a r e exchanged. c a p a c i t y needed. i n S e c t i o n A-3. Real-time compressed speech may b e s e n t by d i v i d i n g i n r e a l time w i t h , f o r example, phase-locked amount of time loops.
I n t e r r u p t e d samples may be j o i n e d and s e n t as d i s c u s s e d
The d i v i d e d frequency i s s e n t f o r t h e This i s a c t u a l l y a form of " s y n t h e s i s -

-- n o t

cycles
t h a t i t o c c u r s and i s multiplexed back
up f o r t h a t time a t t h e r e c e i v e r.
a n a l y s i s " compression where a frequency i s encoded i n t o l / n times i t s e l f and decoded by m u l t i p l y i n g by n a t t h e r e c e i v e r. This t y p e of p r o c e s s i n g goes beyond m o d i f i c a t i o n of t h e v o i c e s i g n a l and w i l l be d i s c u s s e d f u r t h e r i n S e c t i o n B-2, Harmonic Compression. s i o n i s l i m i t e d t o about 3 t o 1.

s t r e t c h - o u t f a c t o r would b e used f o r t h e in-mine t r a n s m i t t e r and a
small s t r e t c h - o u t f a c t o r would be used f o r t h e s u r f a c e t r a n s m i t t e r.
This d i f f e r e n c e i n stretch-out f a c t o r i s p o s s i b l e because t h e power o f s u r f a c e t r a n s m i t t e r i s n o t a s l i m i t e d a s t h a t f o r t h e in-mine t r a n s m i t t e r , b o t h from t h e s a f e t y a s p e c t and t h e a v a i l a b i l i t y of raw power. There a r e a number o f ways t h a t such s t r e t c h e d - o u t v o i c e messages c o u l d be handled. duration. F i g u r e 12 i l l u s t r a t e s some of t h e p o s s i b i l i t i e s.
t h e t o p of t h i s f i g u r e i s shown an i n i t i a l v o i c e message of 2-seconds The second l i n e i l l u s t r a t e s t h e s t r e t c h e d - o u t v e r s i o n of t h i s

1. Original Message

2. Stretched-Out Version of Original Message
3. Received Reconstructed Message
4. Alternate Message Reconstruction at '/z Original Articulation Rate
5. Repeat of Message Reconstructed a t 1/2 Original Articulation Rate

- - - --

Seconds

FIGURE 12

FORMATS FOR RECONSTRUCTION OF STRETCHED-OUT MESSAGES AT 6 TO 1 STRETCH-OUT RATIO
message, t h e t i m e s t r e t c h - o u t f a c t o r being 6 t o 1. The generation of t h i s stretched-out v e r s i o n can s t a r t immediately; i t does n o t r e q u i r e t h a t t h e f u l l message be completed before transmission of t h e s t r e t c h e d out v e r s i o n s t a r t s. There a r e a number of a l t e r n a t i v e ways t h a t t h e r e c e i v e r end can be handled. Perhaps t h e most obvious i s compressing t h e received message This mode i s shown on l i n e 3. by 6 t o 1, such t h a t t h e l a s t p a r t of t h i s reconstruction corresponds t o t h e l a s t p a r t of t h e s t r e t c h e d - o u t version.
A second i n t e r e s t i n g a l t e r n a t i v e i s t o compress t h e received version by
only 3 t o 1, but keeping t h e p i t c h t o a n a t u r a l one ( t h i s i s a f a m i l i a r technique f o r t h e slowed-down speech systems promoted f o r l e a r n i n g f o r e i g n languages from tape recordings) pressed waveform. on l i n e 4.
This compression would be done by
r e p e a t i n g on a subphonemic l e v e l s h o r t passages of t h e received and comThe output presented a t t h e r e c e i v e r would occupy This mode i s shown
4 seconds and end a s t h e stretched-out version ends.
r e p e a t i t twice, a s i l l u s t r a t e d on l i n e 5.
A s t i l l f u r t h e r p o s s i b i l i t y i s t o u s e 3 t o 1 compression and
It i s i n t e r e s t i n g t o n o t e
t h a t t h i s mode r e s u l t s i n received messages s t a r t i n g t o reach t h e o p e r a t o r only 4 seconds a f t e r t h e s t a r t o f transmission. Stretched-out transmission systems seem t o o f f e r one of t h e few v i a b l e means of reducing t r a n s m i t t e r power requirements f o r w i r e l e s s through-the-earth organized. voice communication l i n k s.

. Greefkes and F. DeJaeger , "Continuous D e l t a ~ o d u l aion," t

Philips

Research Report NO. 22, pp 233-246,
APPENDIX I N R O BAND VOICE USING FREQUENCY SLICING AR W
It i s a w e l l known f a c t t h a t t h e f u l l - v o i c e band from 200 t o 6100 Hz
i s n o t n e c e s s a r y t o provide completely a c c e p t a b l e v o i c e channel communications. Indeed, a telephone channel whose t r a n s m i s s i o n band i s confined r a t i o , would f a r exceed t h e requirements f o r One of t h e means o f a c h i e v i n g t o 300 t o 3000 Hz i s g e n e r a l l y considered a good v o i c e channel, and, a t adequate signal-to-noise v o i c e communications w i t h i n c o a l mines.
narrow-band v o i c e c o m u n i c a t i o n s i s t o r e s t r i c t t h e t r a n s m i s s i o n band even f u r t h e r than i s done f o r telephone s e r v i c e. A convenient way t o a c c e s s t h e performance t o be expected i s by u s i n g t h e a r t i c u l a t i o n index developed f o r v o i c e t r a n s m i s s i o n systems. The a r t i c u l a t i o n index (AI) ranges from 0 t o 100%. If a system has an

A 1 of

loo%, i t
i n d i c a t e s t h a t a l l i s o l a t e d words spoken i n t o t h e system (A moderate degree of s k i l l Correspondingly, an A 1 of
would be c o r r e c t l y recognized by a l i s t e n e r.
i s expected f o r both speaker and l i s t e n e r. )
50% would mean c o r r e c t i d e n t i f i c a t i o n of spoken words 50% of t h e time. K r y t e r (13y22) and o t h e r s have developed means f o r p r e d i c t i n g t h e A 1 f o r v o i c e channels. The method i s based on bands of e q u a l c o n t r i b u t i o n r a t i o i n each band. Twenty I f t h e s/N If this Between
t o a r t i c u l a t i o n index and t h e s i g n a l - t o - n o i s e
bands a r e used, each c o n t r i b u t i n g a maximum of 5% t o t h e A I.
r a t i o (long-term average s i g n a l power t o average n o i s e power) i s dB o r g r e a t e r , t h e maximum c o n t r i b u t i o n of 5% p e r band i s achieved. S/N r a t i o i s -12 dB o r l e s s , no c o n t r i b u t i o n i s made t o t h e A I. t h e s e two v a l u e s , A 1 c o n t r i b u t i o n depends l i n e a r l y on t h e dB.

ratio in

Thus, i f t h e S/N i n a band i s +3 dB, t h a t band c o n t r i b u t e s 2.5% t o B u s i n g t h i s means, i t i s p o s s i b l e t o e s t i m a t e t h e performance y

the AI.

of any proposed f r e q u e n c y - r e s t r i c t i v e v o i c e channel. The bands of e q u a l c o n t r i b u t i o n t o a r t i c u l a t i o n i n d e x a r e t a b u l a t e d i n Table I.

TABLE 1-1. T E T FREQUENCY B N S OF EQUAL CONTRIBUTION WNY AD T SPEECH INTELLIGIBILITY O Band No. MidFrequency 270 cps Band No. MidFrequency 1740 cps

Limits 200 t o 330 cps

Limits 1660 t o 1830 c p s
From t h i s t a b l e w e can e a s i l y determine t h a t a t e l e p h o n e channel from 330 t o 3200 Hz, f o r example, shows an a r t i c u l a t i o n index of 75% a t b e s t
t h a t i s , i f t h e S / N r a t i o i n a l l bands i s g r e a t e r t h a n 18 dB.

If the

S / N i n a l l bands d e c r e a s e s t o 8 dB ( 1 / 3 of t h e way t o -12 dB), t h e n t h e
A 1 f a l l s t o 50%. For mine communication needs, t h i s v a l u e of A 1 i s
probably more than adequate.
The reason t h i s i s s o can b e seen from
F i g u r e I-1, which shows t h a t f o r an A1 of 50%, a s e n t e n c e i n t e l l i g i b i l i t y n e a r 95% can be expected,
Syllables, Words Or Sentences Understood-Percez
It i s i n s t r u c t i v e t o examine how power requirements can be expected t o vary f o r a frequency-sliced v o i c e communication system,
W e a l s o assume t h a t t h e signal-to-noise

W start e

w i t h t h e awiiuynption t h a t received n o i s e i s f l a t across t h e v o i c e band, r a t i o of proposed systems i s 200 t o 6100 Hz. I n an a c t u a l equalized a c r o s s t h e band of i n t e r e s t :
system, t h i s would probably be done by p r e - e q h a s i s of t h e v o i c e s i g n a l s , Because t h e n o i s e is f l a t a c r o s s t h e band, w e can equate t h e t o t a l n o i s e power d i r e c t l y t o t h e bandwidth of a h y p o t h e t i c a l system, and w e can p l o t A1 a s a f u n c t i o n of s i g n a l power l e v e l f o r such h y p o t h e t i c a l systems, The r e s u l t s of t h i s e x e r c i s e a r e shown i n Figure 1-2 f o r f o u r conditions: f u l l - v o i c e b a d of 200 t o 6100 Hz; a telephone band of 330 t o
Hz; bands 1 through 9 and 1 , t h e bands of l a r g e s t c o n t r i b u t i o n
t o A 1 p e r Hertz of bandwidth; and bands 1 through 5, t h e f i v e bands of l a r g e s t c o n t r i b u t i o n t o A 1 p e r Hertz of bandwidth. It i s obvious from t h e s e p l o t s t h a t i f AI's l a r g e r than 30% a r e d e s i r e d , t h e r e i s no merit whatever i n terms of s i g n a l power r e q u i r e ments t o t h e use of frequency s l i c i n g. Indeed, t h e merit i s found in t h e use of t h e widest p o s s i b l e bandwidth,

code i s e n t i r e l y

FIGURE. 11-1

HANDHELD KEYBOARD AND DISPLAY
a d e q u a t e t o r e p r e s e n t a f u l l a l p h a b e t of c h a r a c t e r s and messages would s i m p l y b e s p e l l e d o u t i n sequence by t h e o p e r a t o r and s u b s e q u e n t l y encoded and t r a n s m i t t e d by t h e u n i t i n a narrow-band t r a n s m i s s i o n scheme. The keyboard on t h i s p o r t a b l e u n i t is i n t e n d e d f o r o n e - f i n g e r o p e r a t i o n ; t h u s , t h e c h a r a c t e r r a t e i s p r o b a b l y l i i n i t e d t o t h r e e o r f o u r p e r second and t h e b i t r a t e i s a p p r o x i m a t e l y 20 p e r second, a number e a s i l y handled by a narrow-band communication system. Received messages c o u l d b e p r e Such a low s e n t e d on t h e o p t i c a l d i s p l a y p a r t of t h e hand-held u n i t s u c h t h a t r e c e i v e d messages are s p e l l e d o u t i n f r o n t of t h e o p e r a t o r. b i t - r a t e system would a p p e a r t o be c o m p a t i b l e w i t h performance a c h i e v e d , by t h e t r a p p e d miner d e t e c t i o n and l o c a t i o n equipment p r e s e n t l y underg o i n g tests by t h e Bureau, and u n i t s s u i t a b l e f o r u s e i n mines s h o u l d
n o t be d i f f i c u l t t o develop.
FLAG SIGNAL SYSTEMS On two o c c a s i o n s when w e were t a l k i n g w i t h p e o p l e about communication
r e q u i r e m e n t s i n c o a l mines, t h e y immediately s u g g e s t e d t h e u s e of f l a g codes as used a t sea f o r d e v e l o p i n g a message s t r u c t u r e c a p a b l e of t r a n s m i t t i n g a wide v a r i e t y of messages w i t h a r a t h e r small number of symbols. N a t u r a l l y , t h e f l a g would b e r e p l a c e d by coded s i g n a l s , b u t t h e t e c h n i q u e o f communicating could b e made t h e same. Unfortunately, i n t h i s instance a look-up book would b e r e q u i r e d a t b o t h t r a n s m i t t e r and r e c e i v e r t o e n a b l e t h e o p e r a t o r t o p u t t o g e t h e r messages and t o i n t e r p r e t r e c e i v e d messages. E. PANTOGRAPH SYSTEM I n c o n v e r s a t i o n s w i t h o t h e r p e o p l e concerned w i t h t h e narrow-band communication s y s t e m s , we found t h e consensus t o b e t h a t t h e Pantograph System used i n some r e s t a u r a n t s t o communicate a w r i t t e n o r d e r from t h e w a i t r e s s t o t h e k i t c h e n might b e f e a s i b l e f o r mine u s e. I n such a sys-

tem, two a n a l o g s i g n a l s can c h a r a c t e r i z e t h e p o s i t i o n on a p i e c e of p a p e r of a pen from which t h e o r i g i n a l w r i t t e n message can b e r e c o n s t r u c t e d. W have n o t given much c o n s i d e r a t i o n t o t h i s a s a v i a b l e means, b u t i t e i s a n i n t e r e s t i n g p o s s i b i l i t y f o r communication.

F Q THE ELECTRONIC PAD

Some of the people with whom we talked suggested that use could be made of an electronic pad which, again, is another way of transmitting a written or printed message from the sender to the receiver. acters, symbols, pictures, or words. In this instance, a stylus would be run across an electronic pad, writing charThese would be electronically Again, it encoded and transmitted for reconstruction at the surface. narrow band. possiblity
could be expected that the system could operate effectively in a fairly
Little further attention was given to this as an alternate
APPENDIX I11 WHISPERED SPEECH The a n a l y s i s / s y n t h e s i s t y p e systems which form t h e major b u l k o f t h e v o i c e bandwidth compression systems r e q u i r e t h a t p i t c h be measured and a d e c i s i o n be made a s t o whether t h e t r a n s m i t t e d p a r t i s voiced o r unvoiced. Data from Flanagan ( I 2 ) show t h a t t h e c h a n n e l c a p a c i t y r e q u i r e d f o r t h e p i t c h and t h e voiced/unvoiced d e c i s i o n i s a p p r o x i m a t e l y 125 bps. There-
f o r e , i f i t were p o s s i b l e t o drop t h e s e two measurements from t h e a n a l y z i n g equipment, a c o r r e s p o n d i n g r e d u c t i o n i n r e q u i r e d c h a n n e l c a p a c i t y would occur. How much o f a s a v i n g s t h i s r e d u c t i o n i n r e q u i r e d c h a n n e l c a p a c i t y a c h i e v e s h a s t o b e r e f e r r e d t o t h e b i t s p e r second r e q u i r e d f o r t h e f u l l c a p a b i l i t y o f t h e system. These numbers r a n g e from 1000 t o 30,000 b p s ,
s o t h a t a t b e s t a 12% s a v i n g s of c h a n n e l c a p a c i t y , and hence, power, could b e a c h i e v e d by t h e a d o p t i o n of t h i s scheme. One might w e l l i n q u i r e a s t o t h e q u a l i t y o r e a s e of d e a l i n g w i t h conv e r s a t i o n s h e l d i n whispered v o i c e
t h e r e s u l t a f dropping b o t h t h e
v o i c e / u n v o i c e d and t h e p i t c h i n f o r m a t i o n i n a n a n a l y s i s / s y n t h e s i s system. The r e p r o d u c t i o n s of speech from t h e c o n t r o l s i g n a l s would b e produced a s whispered speech. While one might b e l i e v e i t t o be d i f f i c u l t t o c a r r y on a c o n v e r s a t i o n i n whispered speech, t h i s p e r c e p t i o n i s probably b a s e d on t h e d i f f i c u l t y of a r t i c u l a t i n g w i t h whispered s p e e c h , b u t i n t h e systems w e a r e c o n t e m p l a t i n g t h e i n p u t would b e i n normal v o i c e and t h e a n a l y s i s / s y n t h e s i s equipment would be used t o c r e a t e t h e whispered speech. Bernald Gold of M.I.T.'s L i n c o l n Lab, i n a p r i v a t e communication, t o l d u s

By c a p i t a l i z i n g on t h e i n s t a n -
taneous amplitude d i s t r i b u t i o n o f speech, i t o f f e r s a g r e a t economy i n For example, t o p i c k a range on t h e D l curve, i f we want t o m a i n t a i n a 1 7. 5 dB SIN r a t i o a t a s i g n a l power 40 dB below f u l l l o a d w i t h l i n e a r PCM, 1 d i g i t s would be needed p e r sample i n s t e a d 1 of 7. Thus p = 100 companding reduces t h e l i n e b i t r a t e by 36%. A c t u a l P M systems may C
The curves i n F i g u r e I V - 1 a r e i d e a l c u r v e s.
be 1 t o ' 3 dB poorer f o r any of a number of r e a s o n s such a s s m a l l e r r o r s i n sampling time, f i n i t e r a t h e r t h a n i n s t a n t a n e o u s sampling, compressorexpander c h a r a c t e r i s t i c m i s t r a c k i n g , n o i s e i n d e c i s i o n c i r c u i t s , and g e n e r a l system n o i s e. The analog telephone channels d e r i v e d on T systems a r e used i n t e r changeably w i t h a l l o t h e r analog channels i n t h e telephone p l a n t. Thus, i t i s n o t unusual f o r a 1200- o r 2400-bps modem s i g n a l t o be r i d i n g on a T c a r r i e r channel u s i n g 64,000 bps. This d u a l s t a n d a r d poses t a r i f f For example, t o provide a 50i n c o n s i s t e n c i e s f o r telephone companies. r i e r b u t only one from a d i g i t a l. u l a r f a c i l i t y providing s e r v i c e.
kbs d a t a channel, 12 v o i c e channels must b e removed from an analog c a r T a r i f f s a r e n o t geared t o t h e p a r t i c Is t h e f a i r p r i c e f o r a 50-kbs channel
Theoretical D-2 Performance
Signal Power-dB Below Full-Load Sinusoid Source: Ref. 14, 15

FIGURE.IV-1

PCM QUANTIZING NOISE
r o u g h l y 1 o r t i m e s t h a t f o r a v o i c e channel? and Bell's Dataphone D i g i t a l S e r v i c e (DDS). B.

This inconsistency i s

one of t h e p r e s s u r e s t h a t i s l e a d i n g t o a l l d i g i t a l systems s u c h a s D a t r a n
TRANSMISSION O DIGITAL S I G ~ A L SON ANALOG CHANNELS F

- MODEMS

A t t h e same t i m e t h a t B e l l was f i n d i n g i t economically a t t r a c t i v e t o
t r a n s m i t a n a l o g s i g n a l s on d i g i t a l c h a n e l s , t h e i n v e r s e problem a r o s e and grew r a p i d l y. Telephone p l a n t had evolved o v e r t h e y e a r s t o h a n d l e For v o i c e e f f i c i e n t l y and was n o t w e l l s u i t e d f o r raw d i g i t a l d a t a.

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