Owen T. Heisler

Last updated 2019-10-08

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1 Introduction

Please note, this book is a work in progress and very incomplete.

This book is the result of several years of working with speech reinforcement systems, doing research, and a lot of learning by trial-and-error. I do not claim to be an expert, but I do hope that this resource can be helpful for anyone who needs to set up, improve, maintain, or operate sound systems for speech reinforcement.

The primary focus here is on theory, so you can apply it in your particular situation with the hardware you already have. However, hardware choices are very important, so hardware suggestions are also offered. If you would like a ready-to-use complete system design instead, without the theory, please see Design 3.

The goal of a speech reinforcement system is to transport audio signal from a human speaker to the ears of each listener, and to maintain accuracy of the signal as much as possible. The chapters of this book are as follows:

Then there are separate chapters for the following topics:

2 Signal path

The full signal path includes many different parts, and more than just sound system equipment. Each component carrying the signal in the system will affect the signal in some way, often reducing the signal-to-noise ratio (SNR), but usually not raising it. By carefully considering the SNR at each point, it is possible to greatly improve the performance of the system as a whole. There are the three signal paths of interest here:

Human speakermicrophonemixing/processingroom speakersroomears

Human speakermicrophonemixing/processingassistive/outside listeningears

Human speakermicrophonemixing/processingtelephone networkears

3 Human speaker

The human behind the microphone is obviously the most important part of the signal path. A well designed sound system can easily accommodate most voices.

3.1 Suggestions for the person speaking

It is seldom practical to try altering the behavior of the person speaking, and generally impossible to affect a change in the actual sound produced. However, there are a few suggestions that could be shared with any speaker who is interested.

  1. “Speak up!” If the speaker is aware that additional effort is needed, it can actually cause involuntary changes in speech for improved clarity (Lombard effect), in addition to a simple signal-to-noise ratio increase at the microphone.

  2. It can be counter-productive for the lips to be closer to the microphone than about 3 inches. This is caused by the proximity effect, which increases low frequency gain and thus reduces the signal-to-noise ratio. The speaker should move closer to the microphone only when the system has feedback or gain problems (shouldn’t be necessary with correct system design).

3.2 Frequency ranges

Having covered some suggestions for the speaker, we now are concerned with the technical characteristics of voice so we can understand what our “signal” actually is.

Sources:

3.3 Speech critical bands

The following is a selection of narrow frequency bands which may be treated as the elementary signals in speech. This is from Gdansk University of Technology Multitask Noisy Speech Enhancement System.

Speech critical bands
Center Width Range (Hz)
200 60 170-230
300 60 270-330
500 60 470-530
800 70 765-835
1000 80 960-1040
1500 100 1450-1550
2000 130 1935-2065
3000 200 2900-3100
5000 300 4850-5150
8000 600 7700-8300

Consequently, on a 1/3 octave equalizer, the following bands are especially important:

Interestingly, all other bands can be reduced quite a lot without noticably impacting audio quality. However, I do not recommend doing so for normal operation, as everything from 80 Hz to 10 kHz is important for high-fidelity speech reproduction.

4 Microphone

The microphone is the first component in the signal path that you have full control of. You need a good one, one designed specifically for speech. Other traits include: type, pickup pattern, frequency response, and proximity effect.

4.1 Consider these microphones

4.2 Microphone types

Sources:

4.3 Polar (pickup) patterns

Sources:

4.4 Microphone stands

When a microphone and stand have been set up for use, to accommodate most heights, the microphone capsule should be adjustable from 4 ft (120 cm) to 6 ft (180 cm).

I recommend the On-Stage MS7700B Tripod Base Mic Stand as a basic microphone stand.

Here are some portable stands to consider:

Remote motorized stands:

4.5 Notes

5 Mixing/processing

5.1 Feedback suppressors

5.2 Equalizers

5.3 Mixers

5.4 Notes

6 Room speakers

6.1 Selecting speakers

6.2 Placing speakers

6.3 Notes

7 Assistive/outside output

Consider the following requirements:

7.1 Options

Notes:

Sources:

7.2 Notes

8 Telephone

This chapter covers the use of a phone and a conference calling service provider to share the audio from a sound system to remote listeners.

Note that there are also many other ways to implement audio streaming, especially over the internet (such as with Asterisk and MusicOnHold streaming).

8.1 Connecting an output to a phone input

I know of a few products that may be helpful for this. I recommend using a simple wired connection that does not require any power, if possible. However, there are some options I know of that use Bluetooth.

kV Connection KM-IPHONE-MICX*

kV Connection sells two XLR to TRRS adapters for connecting a system output to a cell phone’s microphone input: the KM-IPHONE-MICX and the KM-IPHONE-MICX-A22. These products are identical except that the second one attenuates the signal by 25 dB.

Most outputs will be line level rather than microphone level, and in that case the KM-IPHONE-MICX-A22 is the appropriate option. If the signal level is still too high, or to use the KM-IPHONE-MICX with a line level output, add the Audio-Technica AT8202 adjustable in-line attenuator (selectable 10, 20, or 30 dB attenuation).

As of 2019-05-14, the kvconnection.com website is unavailable. The kV Connection products may no longer be available. If you have any more information, please let me know.

JK Audio BlueDriver-F3

This is a small device that bridges XLR and Bluetooth. It has an internal Li-Ion battery that must be charged via a mini USB port. I used this for a while and found that repeated pairing was unreliable and the single button control was not intuitive (did not seem to match the description in the documentation). These are the steps that I wrote for using this device (which probably differ from the product’s documentation)—do these with the USB cable disconnected:

  1. To pair, start with the device off (see step c.). Hold the Connect button. The blue will light start blinking rapidly, then stay on, than blink once. Now release. The blue light should be blinking steadily. Initiate the pairing process from the other device. Once paired, the device will be ready for use.

  2. To turn on, hold the Connect button. Wait until the blue light starts blinking rapidly, then release.

  3. To turn off, hold the Connect button. The blue light will turn on, blink, and blink again. Now release.

JK Audio Daptor Three
This is similar to the BlueDriver-F3 but supports 2-way audio and uses a standard 9-volt battery.

8.2 Using FreeConferenceCall.com

This service is no longer recommended because it provides free conferencing through the use of traffic pumping. This can result in unexpected blocked calls, dropped calls, and usage charges. See related FCC and T-Mobile pages. Unfortunately I have not found an alternative; please let me know if you have any ideas.

Dial strings in this section use a p to indicate a pause, sometimes dial string pauses are indicated by commas (,). You should be able to save the entire dial string, including pauses, to your phone’s contacts database. Always listen while your phone is dialing to verify successful connection and mode selection.

  1. Register at FreeConferenceCall.com.
  2. Record your dial-in number, access code, and host PIN.
  3. To provide program audio to the conference, use this dial string:
    <dial-in number> pp <access code> #pppp <host PIN> #ppp*5p*5p*8
    (The *5*5 switches to “presentation mode”, with all guests muted; the *8 disables the tones indicating when guests enter or exit the conference.)
  4. Guests can be given the dial-in number and access code; they will be prompted to enter the access code and then press the pound/hash button (#). Here is a dial string:
    <dial-in number> pp <access code> #.

9 Room

The room greatly affects the audio signal. This can be fixed, in part, by installing acoustic treatment. It can be accomodated, in part, by applying to the signal an inverse of the room’s frequency response.

Room acoustics, modes, and responses are perhaps the most complicated part of the signal path. This page is certainly not comprehensive, but aims to touch on some basics.

9.1 Acoustics

Sources:

9.2 Notes

10 Ears

10.1 Frequency

Sources:

10.2 Delay

Sources:

10.3 Notes

11 Wiring and cables

11.1 Mic/line level cable

Use a 2 conductor cable with a shield for any of the following:

Use eg. Belden 9145 for installed wiring (in a building or rack cabinet).

11.2 8 ohm speaker wire

As speaker wire, use 2 conductor wire. Shielding is not necessary and increases the risk of a short.

11.3 70V speaker wire

For 70V, use 2 conductor wire. Shielding is not necessary and increases the risk of a short.

18 AWG wire will yield less than 5% power loss up to about 100 m (330 ft).

For installed 70V audio, use 18 AWG or larger wire. Solid and stranded wire are both acceptable, up to 12 AWG. Stranded may be easier to work with when soldering connectors, especially with large gauge wire.

For portable 70V audio, use stranded 18 AWG wire or larger. Stranded wire is more durable for handling.

11.4 Attenuators & isolators

11.5 Compatible wiring scheme

Amplifier output is 70V, 2 channels, with male twist connector.

Supported outputs:

Standard cables/adapters required:

Special cables/adapters required:

Suggested connectors:

11.6 More speaker wiring

There are two options for speaker wiring: low impedance and high impedance. Low impedance is typical for basic systems with few speakers and short cable runs. High impedance or constant voltage wiring (typically 70V in the United States) requires a transformer at each speaker but offers numerous advantages over low impedance:

Sources:

11.7 Notes

12 Power

12.1 Uninterruptible power supplies (UPSes)

I recommend the CyberPower OR700LCDRM1U.

The following models are similar: OR500LCDRM1U, OR1000LCDRM1U, OR1500LCDRM1U, PR750LCDRM1U, and PR1000LCDRM1U.

13 Storage

For portable storage, I recommend the Gator Cases Pro Series Molded Racks, available from 2U to 12U. The G-PRO-6U-19 case is used in the Design 3 build.

Other options include:

I do not recommend rolling racks due to the potential for damage to hardware caused by shock from rolling across uneven ground (such as cracks in sidewalks).

14 Hardware

When selecting hardware, look for quality first, then look for a feature set close to what you need. Often a well-designed device with a limited feature set will work better than a poorly-designed device with an extensive feature set. Do not make the mistake of looking for the cheapest product with the features you need: likely those features are poorly implemented.

If price is a concern, consider buying used equipment. Generally, used high-quality hardware will serve you better than new low-quality hardware. But remember to test used hardware thoroughly before using in an important event!

Here are some brands I have at least some experience with. You can learn more by testing devices and doing research.

15 Measurements

Being able to take measurements is very important, as it removes much of the guesswork involved in making a sound system effective.

You should be able to do the following measurements:

15.1 Hardware

Audio analyzers are available that can provide much of this functionality. However, the following hardware acts as additional audio inputs/outputs for a computer (via USB) and can be used not only for room measurement, but also for signal processing in general. These been selected for some combination of portability (size), versatility, and quality.

You will also need a measurement microphone. Dayton Audio sells three different test measurement microphones, and provides a unique calibration file for each microphone (cross-referenced by serial number)—this can be used to set a calibration curve for compensation. I recommend the EMM-6. I have also used the dbx RTA-M microphone.

15.2 Software

15.3 Other things of interest

16 Other resources

See the following external links for more information. Some of the links represent content that could be added at some point to the above chapters.

16.1 General

16.2 Speech

16.3 Microphones

16.4 Processing

16.5 Amplifiers and speakers

16.6 Room

16.7 Telephone

16.8 Basic Hardware

16.9 Hardware processors