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     SOFTWARE-BASED DIGITAL AUDIO ON PCs
          by David T. Chappell

Copyright (C) 1991  ACM.  All Rights Reserved

This article originally appeared in the Proceedings of the 1991 ACM
Computer Science Conference, March 1991.  Copying is by permission of
the Association for Computing Machinery.


ABSTRACT

    Digital audio techniques were investigated with special
concentration on computer applications.  Experimentation showed that
the Intel 8253 programmable interval timer, found on all IBM PCs and
compatibles, can output digitized sound.  The relationship between
pulse amplitude modulation and pulse width modulation signals was
found to be significant to this process.    Feeding digitized sound
data to the chip results in an inherent transfer from pulse amplitude
modulation to pulse width modulation encoding.  The relationship in
signal modulation allows IBM PCs to play digitized sound without the
use of extra hardware.

Biography:
    David T. Chappell is a senior in the computer science curriculum
at North Carolina State University.  In addition, he participates in
the co-op program and works for IBM.  His interests are in the areas
of advanced input/output, technological research, and scientific
applications.  David plans to attend graduate school in computer
science or a related field.


INTRODUCTION
    In the past few years, the computer industry has been slowly
gaining interest in computer-generated speech and sound.  In this
area, the Apple Macintosh and Commodore Amiga provide built-in sound
control hardware so that their machines can play digitized recordings
with little effort from the programmer.  IBM, however, has chosen not
to include advanced sound capabilities in its line of personal
computers.  The market has shown that relatively few people will buy
speech or sound add-ons from either IBM or third parties.  Until such
hardware is standardized, sound software will encounter great
difficulty in gaining acceptance.  It is in this regard, however, that
mathematics, engineering, and software can come to the rescue:  it is
possible for a PC to play good quality sound without additional
hardware.


BACKGROUND
    Sound is transmitted through the air as a longitudinal wave.  The
source of the sound compresses the air in one area and this air
compresses the air next to it while it moves back to its original
position.  As each air molecule is displaced from and returns to its
normal position, the wave travels through the air.  The movement that
results from these repeated compressions and rarefactions is called
the propagation of the wave [1].
    It is often convenient to use graphs to visually represent sound.
Most pictures show a graph of the wave amplitude vs. time, where the
amplitude represents the displacement of molecules from their original
positions.  [See figure 1.]  Regular waves, such as sine waves, need
not be depicted in such a manner but can be represented by just a
frequency (or wavelength), amplitude (volume), and duration.  For
example, the SOUND statement in Microsoft BASIC has the parameters of
frequency and duration.  (Volume is not needed because the PC has only
one volume.)  The irregular waves that make up most of the sound we
hear are more complicated and need to be represented by giving
specific details on the amplitude at each point in time.
    Digitized sounds consist of the numerical values for the
amplitude at regular time intervals.  [See figure 2.]  Along with the
amplitude data, the number of samples taken per second must be
recorded.  By reproducing the amplitude changes at the original rate,
the sound can then be played back.  Much research has gone into
digitized sound, and it is now possible to produce digitized
recordings which, to the human ear, are identical to their analog
counterparts.
    Once recorded, digital signals can be stored in a variety of
formats.  Pulse amplitude modulation (PAM) is the standard method of
representing analog data.  [See figure 3.]  In this method, each piece
of data represents the amplitude at one instant in time.  Pulse width
modulation (PWM) treats each piece of information as the duration of a
pulse which starts at a regular frequency.  [See figure 4.]  In pulse
code modulation (PCM),  each bit records whether the amplitude is high
or low at each point in time.  [See figure 5.]  PCM is the most common
method in digital audio recording and is used by audio compact discs
to achieve high-quality sound.  Pulse position modulation (PPM)
records the position of a brief pulse by giving the time duration
before the pulse occurs [1].  [See figure 6.]
    Several brands of personal computers include dedicated sound
chips.  The Macintosh, Amiga, Atari ST, and other computers can easily
produce high-quality sound by using dedicated hardware.  On these
computers, many programs are enhanced by the addition of sound and
speech.  Likewise, IBM and a multitude of other companies have
produced boards that allow PCs to record and playback digitized sound.
A lack of standards, however, limits the use of these PC boards, and
few programs use them.
    Over the years, several attempts have been made to play digitized
sound on PCs without special hardware.  A number of these have
appeared in the public domain, but few are coherent.  Commercial
software has had greater success, but most of these programs produce
rough speech.  With the recent rise of interest in audio and video, a
few developers have even produced good, intelligible speech and
sounds.  The work presented here rivals that of the best public domain
and commercial successes.


MATERIALS AND METHODS
    The Intel 8253 programmable interval timer, found on all IBM PCs
and compatibles, is a flexible counter chip.  It has three 16-bit
channels.  Each channel produces an output signal based on an input
signal and a programmed 16-bit number.  The chip's six modes can
produce varying types of output.  Table 1 summarizes the six modes.
Refer to Rosch [2] or Sargent and Shoemaker [3] for more details.




Table 1:  Intel 8253 Operating Modes

            0 - interrupt on terminal count
            1 - programmable one-shot
            2 - rate generator
            3 - square-wave generator
            4 - software-triggered strobe
            5 - hardware-triggered strobe

    On IBM PCs, the 8253's first channel increments the time-of-day
clock, the second channel refreshes the DRAMs, and the third channel
sends sound to the speaker.  The timer/counter of AT class machines is
based on the Intel 8254-2 chip and is functionally equivalent to the
8253.  The PS/2 line has an 8253 that is used the same way, except
that a separate chip refreshes memory, and the second channel of the
8253 is used for diagnostics or is unassigned [2].
    The 8253 can only give two possible output states:  0 and 1.
Possibly because of this limitation, the chip's use for sound
production in PCs has typically been limited to being a square-wave
generator (mode 2).  When functioning as a square-wave generator, the
chip's output is equal to its input frequency (1.193 MHz on PCs)
divided by a 16-bit number that is input to the chip by the
programmer.  The output is a square wave whose high and low periods
are equal.  The range of the chip falls between 18.2 hertz and 1.193
megahertz [4].  Mode 2 is used by channel 0 for the time-of-day clock
and by channel 2 to produce the beeps and whistles found in many DOS
programs.
    Despite the two-state output limitation, the 8253 can also play
digitized sound.  When put in mode 0, the output defaults to high.
When a 16-bit number is then programmed into the chip, the output goes
low for the duration of the specified number of input pulses, after
which the output returns high.  The net result is that the output wave
is in the form of PWM.  [See figure 7.]
    To program PWM sound output on the 8253, several steps must be
taken.  First, the programmable peripheral interface (PPI) chip must
be initialized to the desired mode.  Then the 8253 must be initialized
and data sent to it.  Table 2 summarizes chip ports.

Table 2:  Chip I/O Port Addresses

            Port                Dec    Hex
            PPI chip            97    61
            8253 Channel 0        64    40
            8253 Channel 1        65    41
            8253 Channel 2        66    42
            8253 Control Word    67    43

    To allow the 8253 to control the speaker, the PPI must be set
correctly.  The two lowest bits of port 97 must be turned on.  The
other bits of this port should remain untouched since they are used
for other purposes [4].
    Port 67 is the control word register which initializes the 8253.
Each counter is initialized by sending this port one control byte.
Table 3 shows the meanings of the bits in the control word.

Table 3:  Meaning of 8253 Control Word Register

        bit 0    = 0, count in binary
                = 1, count in Binary Coded Decimal
        bits 1-3    = mode number (0 to 5 in binary)
        bits 4,5    = 00, latch current count for reading
                = 01, read/load low byte
                = 10, read/load high byte
                = 11, read/load low byte, then high byte
        bits 6,7    = counter number (0 to 2 in binary)

    (Bit 0 is least significant; 7 is most significant)

When reading both bytes of the 16-bit value, a latch command prevents
the count from changing between reading the high byte and the low
byte.  Latching is not needed when reading only a single byte.  For
example to set the chip to generate musical tones in mode 3 the
control word is 182 (B6 hex).  For digital audio via mode 0, use 176
(B0 hex).
    Ports 64, 65, and 66 are used to read and write to timers 0, 1,
and 2 respectively.  Data sent to these ports becomes the 16-bit
number used to affect output.  If only one byte is sent, the other
byte retains its previous value.
    Listing 1 shows the general algorithm for 8-bit digital audio
output with the 8253.  8-bit quality is achieved by leaving the high
byte constant and sending data only to the low byte.


Listing 1:  Algorithm for Digital Audio

            Load Digital Audio Data;
            value = InPort(61h);    -- Initialize PPI
            OutPort(61h, value OR 3);
            OutPort(43h,B0h);        -- Initialize 8253
            OutPort(41h,00h);
            OutPort(41h,00h);
            OutPort(43h,90h);
            loop until end of data    -- Play sound
                OutPort(41h,Data);
                Wait Until Data Passes;
            OutPort(43h,B6h);        -- Restore 8253

    Listing 2 gives an example program written for Turbo C.  In the
sample program, the user must specify whether the input data is
signed.  Some forms of digital audio storage, including PCM,
inherently use unsigned variables.  Other forms, including PWM and
PAM, can be stored as signed variables, and notably the Amiga
microcomputer stores digitized sound data on a scale from -128 to 127.
Since the playback method using the 8253 must use unsigned data, a
scaling factor of 128 must be added to all signed data [5].


Listing 2:  Turbo C Code for Digital Audio

#include <conio.h>
#include <dos.h>
#include <io.h>
#include <stdio.h>
#include <stdlib.h>

/*    SOUND.C

    Author:   David Chappell
    Version:  1.46d
    Date:     24 June 1990
    Method:   8253 PWM method
*/

FILE *soundfile;    /* input data file */
unsigned long size;    /* size of input file */
int wait,      /* time to wait between sending samples out */
unsigned char  offset,  /* change signed samples to unsigned */
                vol1, vol2;  /* adjusts range of data */

void error(char message[])
/* Purpose:  handles errors */
    {
    fprintf(stderr,"\nERROR:  %s\n",message);
    exit(-1);
    }

void playfile(void)
/* Purpose:  loads file and plays digitized sound */
    {
    int pause;
    unsigned int count, temp;
    unsigned char min, max, data;
    char *inputbuffer;

    if ((inputbuffer=(char*) calloc(size,sizeof(char))) == NULL)
        error("Not enough memory to load file");
    fread(inputbuffer, size, 1, soundfile);
    /* scale data */
    min=255;  max=0;
     for (count = 0; count < size; count++) {
        data = *(inputbuffer + count)+offset;
        if (data<min)
            min=data;
        if (data>max)
            max=data;
        }
    vol1 = 64;
    vol2 = max-min+1;      /* scale from 0 to 64 */
    offset -= min;         /* move lowest point to zero */
    disable();        /* disable interrupts */
    for (count = 0; count < size; count++) {
        data = *(inputbuffer + count) + offset;
        temp = data * vol1 / vol2;
        data = temp + 1;
        output(66,data);
        for (pause = 0; pause < wait; pause++);
        }
    enable();            /* enable interrupts */
    }

void startspeaker(void)
/* Purpose:  intitialize speaker for output */
    {
    outp(97,inp(97) | 3);    /* set PPI */
    outp(67,176);        /* send initial data to timer */
    outp(66,00);
    outp(66,00);
    outp(67,144);        /* prepare timer chip to receive data */
    }

void openfile(void)
/* Purpose:  gets information from user and opens input file */
    {
    char choice;        /* key hit by user */

    clrscr();            /* clear screen */
    puts("What file do you want to hear?");
    if ((soundfile = fopen(gets(NULL),"rb")) == NULL)
       error("Unable to open sound file");
    fseek(soundfile,0,SEEK_SET);
    size = filelength(fileno(soundfile));
    printf("\nFile size = %lu bytes\n\n",size);
    printf("What delay time do you want in FOR counter? ");
    scanf("%d",&wait);
    printf("Is the data signed? ");
    choice=getche();
    if ((choice=='Y') || (choice=='y'))
       offset=128;
    }

void stopsound(void)
/* Purpose:  resets speaker to stop sound */
    {
    outp(67,182);        /* restore timer to mode 3 */
    fclose(soundfile);
    }

void main(void)
   {
   openfile();
   startspeaker();
   playfile();
   stopsound();
   }

RESULTS
    The method described thus far has several limitations when put
into practice on PCs.  The 16-bit quality of the chip reduces to 7
bits at most sample rates.  Also, a background tone is produced along
with the desired sound because of the use of PWM.  Both of these
difficulties arise from timing problems.
    The method described thus far has the capability of yielding
high-quality 16-bit sound.  The code given, however, can only produce
approximately 7-bit sound.  Although the 8253 has the ability to play
16-bit data, the timing limitations of the PC restrict the length of
each pulse.  In order to produce sound at the rate of about 8-13 kHz,
only about six or seven bits of data are processed before the next
piece of data must begin output.  At a slower sample rate of about 4-7
kHz, seven to eight bits of accuracy can be achieved.  A higher input
frequency would resolve this difficulty; however, the frequency can
not be changed in PCs but could be modified in other applications of
the 8253.
    The maximum data size (volume) possible can be calculated
mathematically.  The 8253 input rate divided by the output sample rate
yields the number of time periods that pass before the next sample
begins play:

    Maximum value = 1.193 MHz / sample rate
For example, the maximum volume for an 8 kHz sample is 147.  The 1.193
MHz input frequency that feeds the 8253 limits the chips sound
capabilities.
    As an annoying side-effect, the provided algorithm creates a
background tone.  Due to the nature of PWM, at the beginning of each
piece of data, the output changes state as it goes from high to low.
[See figure 7.]  This periodic oscillation produces a pitch equal to
the frequency at which the sound is played.  For example, an 8 kHz
sample will produce a background tone of 8 kHz.  The resulting tone
overlays the digitized sound output.  The human ear can not detect
pitches beyond 22 kHz, but most people can not hear a pitch of 18 kHz
or greater frequency.  Thus, any sample of this frequency will not
produce an audible background tone.  If a given sample is not of high
enough frequency, this problem can be alleviated by playing each piece
of data multiple times in rapid succession so that the background tone
is of such a high frequency that it is inaudible.  For example, by
playing each datum of a 9.5 kHz sample twice, the resulting pitch will
be 19 kHz.  The first problem, however, becomes dramatic when a
moderate-speed sample is sent repeatedly:  in order to maintain the
original sampling rate, the 8253 has time to process fewer and fewer
bits for each datum [5].


DISCUSSION
    Several other methods were attempted before the above method was
discovered.  Although all of these other methods yield sound of
varying quality, most do not produce recognizable results, and none
match the PWM method.  With some work, any of the following techniques
might be able to give better output.
    One idea is to look at each piece of data, compare it to the
midpoint, and set the speaker bit appropriately.  For example, the
midpoint for 8-bit data is alternately 0 or 128 (depending on whether
a sign bit is used), so any values above 128 (or 0) could be set as 1
and all other values set as 0.  Modifying bit 3 of I/O port 97 (61
hex) changes the state of the speaker's position.  This idea could
never yield very good results, however, as it gives only 1-bit
accuracy.
    A heretofore ineffective method is based on PCM encoding.  Each
individual bit of input directly determines the state of the speaker.
Manipulating bit 3 of port 97 yields the two states.
    Another possibility is to use mode 3 of the 8253 to play a pitch
which is either directly or inversely proportional to the data.
Although this method yields audible sound, the results are not as good
as those of the main PWM method.
    A final idea is to use the 8253 based on pulse position
modulation.  Mode 4 of the 8253 should produce PPM output, but trials
so far have given negative results.

    The given algorithm can be further improved by allowing the
program to automatically determine the speed of the computer.  In the
current method, a delay time must be entered manually.  By
incrementing a counter while watching the system clock, a ratio of
instructions to time can be calculated.  Under PC-DOS and OS/2, the
18.2 Hz frequency of the system clock limits the accuracy of this
idea, however, unless channel 0 of the 8253 is used to change the
system timer.
    A better timing method would be to use an algorithm that does not
rely on loops for timing delays.  Under PC-DOS, interrupts can be
modified so that data is played after each clock tick.  By increasing
the frequency of clock ticks and changing the timer interrupt, the
computer can output each piece of data at a regular interval.  Other
operating systems, however, do not allow such interrupt modification.

    The development of this digital audio playback method has several
implications and possibilities.  A variety of applications, from games
to word processors, can use voice and sound.  On multi-tasking
operating systems, sounds can easily be played in the background.
Other computers could use the same ideas for audio output.  When
combined with extra hardware, this method can form a complete audio
I/O system.  When a PC acts as a terminal, this procedure will allow
mainframes and minicomputers to play digitized sound.
    Speech interaction is currently put to several uses.  For
example, IBM's SpeechViewer helps deaf children and adults improve
pronunciation.  In addition, several programs use spoken words to help
people learn to read and write.  For blind users, IBM's ScreenReader
program can vocally relate the text that appears on the monitor.
There are numerous instances of disabled users benefitting from
talking computers.
    In addition, many musicians use computers to produce and mix
sounds.  Computers can now produce rich tones that rival musical
instruments.
    By reducing the need for extra hardware, digitized sound can
easily be added to other programs.  As an obvious example, games can
use sound for both special effects and general entertainment.  Useful
applications, from word processors to spreadsheets, can speak to help
visually impaired users.  Inexperienced users would find a computer to
be much friendlier if it could speak to them.  A user interface that
includes speech can help bring computers to the level of interaction
that humans use with each other.  Thus, nearly all types of software
can benefit from the addition of speech and sound capabilities [5].
    When running under PC-DOS, only one program can be run at a time.
The only way to allow the computer to play recorded sounds while
continuing other work is to modify interrupts, as mentioned above.
Under multi-tasking operating systems, such as OS/2 and Unix, the 8253
could be continually fed data in a background task while the main
program continues.  Playing sound in the background gives more
flexibility.  For example, a communications package could verbally
report an error while continuing to receive data, or a graphical demo
could play music in the background while displaying picture on the
monitor.
    The data storage method used here is compatible with many others.
A huge number of digitized samples are available from Macintosh,
Amiga, and Atari ST computers.  Several PC expansion boards also use
the same storage method.  Data recorded on any of this hardware can be
played back on an ordinary PC.  Whether the original sample is in the
form of PAM, PCM, or PWM, it can play through the PC speaker.
Furthermore, by purchasing an available expansion board or building
one, sound recording is possible on PCs.
    As the use of speech technology grows, speech can be added to
larger machines.  According to IBM's long-range plan, all host
computers will eventually be accessed via a PS/2 running OS/2.
Although mainframes and minicomputers do not typically have sound
capabilities, they could be use the PS/2's speaker for speech output.
Thus, by using PCs as terminals, the full range of computers can
handle digital audio.
    The algorithm presented here can be used in settings other than
in a PC.  The same method could be used in any computer with an 8253
chip, and a hardware expansion using the 8253 can be added to other
computers.  More importantly, any hardware configuration capable of
producing output similar to PWM can, when connected to a speaker,
produce digitized sound.  Similarly, any system able to produce pulses
similar to any digital recording method can output digitized sound.
As a result, hardware with only two output states can play sound, and
a digital-to-analog converter, such as found in the Amiga, is not
required.


CONCLUSION
    Over the past several decades, engineers have searched for ways
to make computers both talk and play high-quality music.  One solution
to both problems, digitization of sound using pulse modulation,
requires little processing time and is thus appropriate for
microcomputers.  Although previous usage of digitized sound has been
limited to computers with specialized hardware, it is possible for a
standard PC to play good quality sounds without extra hardware.  As
the computer world strides deeper into multimedia and other sound-
based applications despite a lack of hardware standards for sound
output on PCs, this method may prove to be invaluable in bringing
sound to the masses.  With minimal effort, any program can add a new
dimension with speech, music, and sound effects.


REFERENCES


[1] Pohlmann, Ken C.  Principles of Digital Audio.  H. W. Sams,
Indianapolis (1985).

[3] Rosch, Winn L.  The Winn Rosch Hardware Bible.  Simon & Schuster,
New York (1988).

[5] Sargent, Murray, III and Richard L. Shoemaker.  The IBM Personal
Computer from the Inside Out.  Addison-Wesley (1984).

[2] Norton, Peter.  The Peter Norton Programmer's Guide to the IBM PC.
1st ed.  Microsoft, Redmond, WA (1985).

[6] Chappell, David T.  "Achieving Inexpensive Digital Audio on PCs
for Educational Purposes".  Proceedings of the Southeastern Small
College Computing Conference.  (1990).



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