Amiga® Hardware Reference Manual: 5 Audio Hardware

This chapter shows you how to directly access the audio hardware to
produce sounds. The major topics in this chapter are:

   *  A brief overview of how a computer produces sound.

   *  How to produce simple steady and changing sounds and more
      complex ones.

   *  How to use the audio channels for special effects, wiring them
      for stereo sound if desired, or using one channel to modulate

   *  How to produce quality sound within the system limitations.

A section at the end of the chapter gives you values to use for creating
musical notes on the equal-tempered musical scale.

This chapter is not a tutorial on computer sound synthesis; a thorough
description of creating sound on a computer would require a far longer
document.  The purpose here is to point the way and show you how to use
the Amiga's features. Computer sound production is fun but complex, and it
usually requires a great deal of trial and error on the part of the
user -- you use the instructions to create some sound and play it back,
readjust the parameters and play it again, and so on.

The following works are recommended for more information on
creating music with computers:

   *  Wayne A. Bateman, Introduction to Computer Music
      (New York: John Wiley and Sons, 1980).

   *  Hal Chamberlain, Musical Applications of Microprocessors
      (Rochelle Park, New Jersey:  Hayden, 1980).

 Introducing Sound Generation      Using Direct (Non-DMA) Audio Output 
 Forming and Playing a Sound       The Equal-tempered Musical Scale 
 Producing Complex Sounds          Decibel Values for Volume Ranges 
 Producing High-quality Sound      The Audio State Machine 

5 Audio Hardware / Introducing Sound Generation

Sound travels through air to your ear drums as a repeated cycle of air
pressure variations, or sound waves. Sounds can be represented as graphs
that model how the air pressure varies over time. The attributes of a
sound, as you hear it, are related to the shape of the graph. If the
waveform is regular and repetitive, it will sound like a tone with steady
pitch (highness or lowness), such as a single musical note. Each
repetition of a waveform is called a cycle of the sound. If the waveform
is irregular, the sound will have little or no pitch, like a loud clash or
rushing water. How often the waveform repeats (its frequency) has an
effect upon its pitch; sounds with higher frequencies are higher in pitch.
Humans can hear sounds that have a frequency of between 20 and 20,000
cycles per second. The amplitude of the waveform (highest point on the
graph), is related to the perceived loudness of the sound. Finally, the
general shape of the waveform determines its tone quality, or timbre.
Figure 5-1 shows a particular kind of waveform, called a sine wave, that
represents one cycle of a simple tone.

 Figure 5-1: Sine Waveform  
Figure 5-1: Sine Waveform In electronic sound recording and output devices, the attributes of sounds are represented by the parameters of amplitude and frequency. Frequency is the number of cycles per second, and the most common unit of frequency is the Hertz (Hz), which is 1 cycle per second. Large values, or high frequencies, are measured in kilohertz (KHz) or megahertz (MHz). Frequency is strongly related to the perceived pitch of a sound. When frequency increases, pitch rises. This relationship is exponential. An increase from 100 Hz to 200 Hz results in a large rise in pitch, but an increase from 1,000 Hz to 1,100 Hz is hardly noticeable. Musical pitch is represented in octaves. A tone that is one octave higher than another has a frequency twice as high as that of the first tone, and its perceived pitch is twice as high. The second parameter that defines a waveform is its amplitude. In an electronic circuit, amplitude relates to the voltage or current in the circuit. When a signal is going to a speaker, the amplitude is expressed in watts. Perceived sound intensity is measured in decibels (db). Human hearing has a range of about 120 db; 1 db is the faintest audible sound. Roughly every 10 db corresponds to a doubling of sound, and 1 db is the smallest change in amplitude that is noticeable in a moderately loud sound. Volume, which is the amplitude of the sound signal which is output, corresponds logarithmically to decibel level. The frequency and amplitude parameters of a sine wave are completely independent. When sound is heard, however, there is interaction between loudness and pitch. Lower-frequency sounds decrease in loudness much faster than high-frequency sounds. The third attribute of a sound, timbre, depends on the presence or absence of overtones, or harmonics. Any complex waveform is actually a mixture of sine waves of different amplitudes, frequencies, and phases (the starting point of the waveform on the time axis). These component sine waves are called harmonics. A square waveform, for example, has an infinite number of harmonics. In summary, all steady sounds can be described by their frequency, overall amplitude, and relative harmonic amplitudes. The audible equivalents of these parameters are pitch, loudness, and timbre, respectively. Changing sound is a steady sound whose parameters change over time. In electronic production of sound, an analog device, such as a tape recorder, records sound waveforms and their cycle frequencies as a continuously variable representation of air pressure. The tape recorder then plays back the sound by sending the waveforms to an amplifier where they are changed into analog voltage waveforms. The amplifier sends the voltage waveforms to a loudspeaker, which translates them into air pressure vibrations that the listener perceives as sound. A computer cannot store analog waveform information. In computer production of sound, a waveform has to be represented as a finite string of numbers. This transformation is made by dividing the time axis of the graph of a single waveform into equal segments, each of which represents a short enough time so the waveform does not change a great deal. Each of the resulting points is called a sample. These samples are stored in memory, and you can play them back at a frequency that you determine. The computer feeds the samples to a digital-to-analog converter (DAC), which changes them into an analog voltage waveform. To produce the sound, the analog waveforms are sent first to an amplifier, then to a loudspeaker. Figure 5-2 shows an example of a sine wave, a square wave, and a triangle wave, along with a table of samples for each.  Figure 5-2: Digitized Amplitude Values
Figure 5-2: Digitized Amplitude Values TIME SINE SQUARE TRIANGLE ---- ---- ------ -------- 0 0 100 0 1 39 100 20 2 75 100 40 3 103 100 60 4 121 100 80 5 127 100 100 6 121 100 80 7 103 100 60 8 75 100 40 9 39 100 20 10 0 -100 0 11 -39 -100 -20 12 -75 -100 -40 13 -103 -100 -60 14 -121 -100 -80 15 -127 -100 -100 16 -121 -100 -80 17 -103 -100 -60 18 -75 -100 -40 19 -39 -100 -20 Note: ----- The illustrations are not to scale and there are fewer dots in the wave forms than there are samples in the table. The amplitude axis values 127 and -128 represent the high and low limits on relative amplitude. The Amiga Sound Hardware

5 / Introducing Sound Generation / The Amiga Sound Hardware

The Amiga has four hardware sound channels. You can independently program
each of the channels to produce complex sound effects. You can also attach
channels so that one channel  modulates  the sound of another or combine
two channels for stereo effects.

Each audio channel includes an eight-bit  digital-to-analog converter 
driven by a direct memory access (DMA) channel. The audio DMA can retrieve
two data samples during each horizontal video scan line. For simple,
steady tones, the DMA can automatically play a waveform repeatedly; you
can also program all kinds of complex sound effects.

There are two methods of basic sound production on the Amiga -- automatic
(DMA) sound generation and  direct (non-DMA)  sound generation. When you
use automatic sound generation, the system retrieves data automatically by
direct memory access.

5 Audio Hardware / Forming and Playing a Sound

This section shows you how to create a simple, steady sound and play it.
Many basic concepts that apply to all sound generation on the Amiga are
introduced in this section.

To produce a steady tone, follow these basic steps:

   1. Decide which channel to use.
   2. Define the waveform and create the sample table in memory.
   3. Set registers telling the system where to find the data and the
      length of the data.
   4. Select the volume at which the tone is to be played.
   5. Select the sampling period, or output rate of the data.
   6. Select an audio channel and start up the DMA.

 Deciding Which Channel to Use          Playing the Waveform 
 Creating the Waveform Data             Stopping the Audio Dma 
 Telling the System About the Data      Audio Summary 
 Selecting the Volume                   Audio Example 
 Selecting the Data Output Rate 

5 / Forming and Playing a Sound / Deciding Which Channel to Use

The Amiga has four audio channels. Channels 1 and 2 are connected to the
left-side stereo output jack. Channels 0 and 3 are connected to the
right-side output jack. Select a channel on the side from which the output
is to appear.

5 / Forming and Playing a Sound / Creating the Waveform Data

The waveform used as an example in this section is a simple sine wave,
which produces a pure tone. To conserve memory, you normally define only
one full cycle of a waveform in memory. For a steady, unchanging sound,
the values at the waveform's beginning and ending points and the trend or
slope of the data at the beginning and end should be closely related. This
ensures that a continuous repetition of the waveform sounds like a
continuous stream of sound.

Sound data is organized as a set of eight-bit data items; each item is a
sample from the waveform. Each data word retrieved for the audio channel
consists of two samples. Sample values can range from -128 to +127.

As an example, the data set shown below produces a close approximation to
a sine wave.

   About the sample data.
   The data is stored in byte address order with the first digitized
    amplitude  value at the lowest byte address, the second at the next
   byte address, and so on. Also, note that the first byte of data must
   start at a word-address boundary.  This is because the audio DMA
   retrieves one word (16 bits) at a time and uses the sample it reads
   as two bytes of data.

To use audio channel 0, write the address of "audiodata" into AUD0LC,
where the audio data is organized as shown below.  For simplicity,
"AUDxLC" in the table below stands for the combination of the two actual
location registers ( AUDxLCH and AUDxLCL ).  For the  audio DMA  channels
to be able to retrieve the data, the data address to which AUD0LC points
must be somewhere in chip RAM.

    Table 5-1: Sample Audio Data Set for Channel 0

     audiodata --->  AUD0LC *        100      98
                     AUD0LC + 2 **    92      83
                     AUD0LC + 4       71      56
                     AUD0LC + 6       38      20
                     AUD0LC + 8        0     -20
                     AUD0LC + 10     -38     -56
                     AUD0LC + 12     -71     -83
                     AUD0LC + 14     -92     -83
                     AUD0LC + 16    -100     -98
                     AUD0LC + 18     -92     -83
                     AUD0LC + 20     -71     -56
                     AUD0LC + 22     -38     -20
                     AUD0LC + 24       0      20
                     AUD0LC + 26      38      56
                     AUD0LC + 28      71      83
                     AUD0LC + 30      92      98
     * Audio data is located on a word-address boundary.
     ** AUD0LC stands for  AUD0LCL and AUD0LCH .

5 / Forming and Playing a Sound / Telling the System About the Data

In order to retrieve the sound data for the audio channel, the system needs
to know where the data is located and how long (in words) the data is.

The location registers  AUDxLCH and AUDxLCL  contain the high three bits
and the low fifteen bits, respectively, of the starting address of the
audio data. Since these two register addresses are contiguous, writing a
long word into AUDxLCH moves the audio data address into both locations.
The "x" in the register names stands for the number of the audio channel
where the output will occur. The channels are numbered 0, 1, 2, and 3.

These registers are location registers, as distinguished from pointer
registers. You need to specify the contents of these registers only once;
no resetting is necessary when you wish the audio channel to keep on
repeating the same waveform. Each time the system retrieves the last audio
word from the data area, it uses the contents of these location registers
to again find the start of the data. Assuming the first word of data
starts at location "audiodata" and you are using channel 0, here is how to
set the location registers:

        LEA     CUSTOM,a0       ; Base chip address...
        LEA     AUDIODATA,a1
        MOVE.L  a1,AUD0LCH(a0)  ;Put address (32 bits)
                                ;  into location register.

The length of the data is the number of samples in your waveform divided
by 2, or the number of words in the data set. Using the sample data set
above, the length of the data is 16 words. You write this length into the
audio data length register for this channel. The length register is called
AUDxLEN, where "x" refers to the channel number. You set the length
register AUD0LEN to 16 as shown below.

        LEA     CUSTOM,a0       ; Base chip address
        MOVE.W  #16,AUD0LEN(a0) ; Store the length...

5 / Forming and Playing a Sound / Selecting the Volume

The volume you set here is the overall volume of all the sound coming from
the audio channel. The relative loudness of sounds, which will concern you
when you combine notes, is determined by the  amplitude  of the wave form.
There is a six-bit volume register for each audio channel. To control the
volume of sound that will be output through the selected audio channel,
you write the desired value into the register AUDxVOL, where "x" is
replaced by the channel number. You can specify values from 64 to 0. These
volume values correspond to decibel levels. At the end of this chapter is
a table showing the  decibel value  for each of the 65 volume levels.

For a typical output at volume 64, with maximum data values of -128 to
127, the voltage output is between +.4 volts and -.4 volts. Some volume
levels and the corresponding decibel values are shown in Table 5-2.

                Table 5-2: Volume Values

          Volume  Decibel Value
          ------  -------------
            64         0         (maximum volume)
            48        -2.5
            32        -6.0
            16        -12.0      (12 db down from the volume
                                  at maximum level)

For any volume setting from 64 to 0, you write the value into bits 5-0 of
AUD0VOL. For example:

        LEA     CUSTOM,a0
        MOVE.W  #48,AUD0VOL(a0)

The decibels are shown as negative values from a maximum of 0 because this
is the way a recording device, such as a tape recorder, shows the
recording level. Usually, the recorder has a dial showing 0 as the optimum
recording level. Anything less than the optimum value is shown as a minus

5 / Forming and Playing a Sound / Selecting the Data Output Rate

The pitch of the sound produced by the waveform depends upon its
frequency. To tell the system what frequency to use, you need to specify
the sampling period. The sampling period specifies the number of system
clock  ticks , or timing intervals, that should elapse between each sample
(byte of audio data) fed to the  digital-to-analog converter  in the audio
channel. There is a  period register  for each audio channel. The value of
the  period register  is used for count-down purposes; each time the
register counts down to 0, another sample is retrieved from the waveform
data set for output. In units, the period value represents clock ticks per
sample. The minimum period value you should use is 124 ticks per sample
NTSC (123 PAL) and the maximum is 65535. These limits apply to both PAL and
NTSC machines. For high-quality sound, there are other constraints on the
sampling period (see the section called  Producing High-quality Sound ).

   The period is inversely proportional to the frequency.
   A low period value corresponds to a higher frequency sound and a
   high period value corresponds to a lower frequency sound.

 Limitations on Selection of Sampling Period 
 Specifying the Period Value 

5 / / Data Output Rate / Limitations on Selection of Sampling Period

The  sampling period  is limited by the number of DMA cycles allocated to
an audio channel. Each audio channel is allocated one DMA slot per
horizontal scan line of the screen display. An audio channel can retrieve
two data samples during each horizontal scan line. The following
calculation gives the maximum sampling rate in samples per second.

   2 samples/line * 262.5 lines/frame * 59.94 frames/second = 31,469

The figure of 31,469 is a theoretical maximum. In order to save buffers,
the hardware is designed to handle 28,867 samples/second. The system
timing interval is 279.365 nanoseconds, or .279365 microseconds. The
maximum sampling rate of 28,867 samples per second is 34.642 microseconds
per sample (1/28,867 = .000034642). The formula for calculating the
 sampling period  is:

                  sample interval     clock constant
   Period value = --------------- = ------------------
                  clock interval    samples per second

Thus, the minimum period value is derived by dividing 34.642 microseconds
per sample by the number of microseconds per interval:

                      34.642 microseconds/sample
   Minumum period = ------------------------------ = 124 timing
                    0.279365 microseconds/interval   intervals/sample


                    3,579,545 ticks/second
   Minumum period = ----------------------  = 124 ticks/sample
                    28,867 samples/second

Therefore, a value of at least 124 must be written into the
 period register  to assure that the audio system DMA will be able to
retrieve the next data sample. If the period value is below 124, by the
time the cycle count has reached 0, the audio DMA will not have had
enough time to retrieve the next data sample and the previous sample will
be reused.

28,867 samples/second is also the maximum sampling rate for PAL systems.
Thus, for PAL systems, a value of at least 123 ticks/sample must be
written into the  period register .

                       Clock Values

                      NTSC        PAL     units
                      ----        ---     -----
   Clock Constant    3579545    3546895   ticks per second
   Clock Interval   0.279365   0.281937   microseconds per interval

   The Clock Interval is derived from the clock constant, where:

               clock interval = --------------
                                clock constant

   then scale the result to microseconds. In all of these calculations
   "ticks" and "timing intervals" refer to the same thing.

5 / / Selecting the Data Output Rate / Specifying the Period Value

After you have selected the desired interval between data samples, you can
calculate the value to place in the period register by using the period

                  desired interval     clock constant
   Period value = ---------------- = ------------------
                   clock interval    samples per second

As an example, say you wanted to produce a 1 KHz sine wave, using a table
of eight data samples (four data words) (see Figure 5-3).

 Figure 5-3: Example Sine Wave  
Figure 5-3: Example Sine Wave Sampled Values: 0 90 127 90 0 -90 -127 -90 To output the series of eight samples at 1 KHz (1,000 cycles per second), each full cycle is output in 1/1000th of a second. Therefore, each individual value must be retrieved in 1/8th of that time. This translates to 1,000 microseconds per waveform or 125 microseconds per sample. To correctly produce this waveform, the period value should be: 125 microseconds/sample Period value = ------------------------------ = 447 timing 0.279365 microseconds/interval intervals/sample To set the period register, you must write the period value into the register AUDxPER, where "x" is the number of the channel you are using. For example, the following instruction shows how to write a period value of 447 into the period register for channel 0. SETAUD0PERIOD: LEA CUSTOM,a0 MOVE.W #447,AUD0PER(a0) To produce high-quality sound, avoiding aliasing distortion , you should observe the limitations on period values that are discussed in the section called "Producing Quality Sound." For the relationship between period and musical pitch, see the section at the end of the chapter, which contains a listing of the equal-tempered musical scale .

5 / Forming and Playing a Sound / Playing the Waveform

After you have defined the audio data  location ,  length ,  volume  and
 period , you can play the waveform by starting the DMA for that audio
channel. This starts the output of sound. Once started, the DMA continues
until you specifically stop it. Thus, the waveform is played over and over
again, producing the steady tone. The system uses the value in the
 location registers  each time it replays the waveform.

For any audio DMA to occur (or any other DMA, for that matter), the
DMAEN bit in  DMACON  must be set. When both DMAEN and AUDxEN are set, the
DMA will start for channel x. All these bits and their meanings are shown
in table 5-3.

         Table 5-3: DMA and Audio Channel Enable Bits

              DMACON  Register

          Bit     Name      Function
          ---     ----      --------
         15       SET/CLR   When this bit is written as a 1, it
                            sets any bit in DMACONW for which
                            the corresponding bit position is
                            also a 1, leaving all other bits alone.

          9       DMAEN     Only while this bit is a 1 can
                            any direct memory access occur.

          3       AUD3EN    Audio channel 3 enable.
          2       AUD2EN    Audio channel 2 enable.
          1       AUD1EN    Audio channel 1 enable.
          0       AUD0EN    Audio channel 0 enable.

For example, if you are using channel 0, then you write a 1 into bit 9 to
enable DMA and a 1 into bit 0 to enable the audio channel, as shown below.

        LEA     CUSTOM,a0

5 / Forming and Playing a Sound / Stopping the Audio Dma

You can stop the channel by writing a 0 into the  AUDxEN  bit at any time.
However, you cannot resume the output at the same point in the waveform by
just writing a 1 in the bit again. Enabling an audio channel almost always
starts the data output again from the top of the list of data pointed to
by the  location registers  for that channel. If the channel is disabled
for a very short time (less than two  sampling periods ) it may stay on
and thus continue from where it left off.

The following example shows how to stop audio DMA for one channel.

        LEA     CUSTOM,a0
        MOVE.W  #(DMAF_AUD0),DMACON(a0)

5 / Forming and Playing a Sound / Audio Summary

These are the steps necessary to produce a steady tone:

   1. Define the waveform.
   2. Create the data set containing the pairs of data samples (data
      words). Normally, a data set contains the definition of one
   3. Set the location registers:

          AUDxLCH  (high three bits)
          AUDxLCL  (low fifteen bits)

   4. Set the length register,  AUDxLEN , to the number of data words to
      be retrieved before starting at the address currently in  AUDxLC .
   5. Set the volume register,  AUDxVOL .
   6. Set the period register,  AUDxPER 
   7. Start the audio DMA by writing a 1 into bit 9,  DMAEN  , along with
      a 1 in the  SET/CLR  bit and a 1 in the position of the  AUDxEN  bit
      of the channel or channels you want to start.

5 Audio Hardware / Producing Complex Sounds

In addition to simple tones, you can create more complex sounds, such as
different musical notes joined into a one-voice melody, different notes
played at the same time, or modulated sounds.

 Joining Tones 
 Playing Multiple Tones at the Same Time 
 Modulating Sound 

5 / Producing Complex Sounds / Joining Tones

Tones are joined by writing the  location  and  length registers ,
starting the audio output, and rewriting the registers in preparation for
the next audio waveform that you wish to connect to the first one. This is
made easy by the timing of the  audio interrupts  and the existence of
back-up registers. The  location  and  length registers  are read by the
DMA channel before audio output begins. The DMA channel then stores the
values in back-up registers.

Once the original registers have been read by the DMA channel, you can
change their values without disturbing the operation you started with the
original register contents. Thus, you can write the contents of these
registers, start an audio output, and then rewrite the registers in
preparation for the next waveform you want to connect to this one.

 Interrupts  occur immediately after the audio DMA channel has read the
 location  and  length registers  and stored their values in the back-up
registers. Once the  interrupt  has occurred, you can rewrite the
registers with the location and length for the next waveform segment. This
combination of back-up registers and  interrupt  timing lets you keep one
step ahead of the audio DMA channel, allowing your sound output to be
continuous and smooth.

If you do not rewrite the registers, the current waveform will be
repeated. Each time the  length counter  reaches zero, both the  location 
and  length registers  are reloaded with the same values to continue the
audio output.

 Audio DMA Example 

5 / / Joining Tones / Audio DMA Example

This example details the system audio DMA action in a step-by-step

Suppose you wanted to join together a sine and a triangle waveform,
end-to-end, for a special audio effect, alternating between them. The
following sequence shows the action of your program as well as its
interaction with the audio DMA system. The example assumes that the
 period ,  volume , and  length  of the data set remains the same for the
sine wave and the triangle wave.

                         Interrupt Program
If (wave = triangle)
               write  AUD0LCL  with address of sine wave data.

Else if (wave = sine)
               write  AUD0LCL  with address of triangle wave data.

                          Main Program

1. Set up  volume ,  period , and  length .
2. Write  AUD0LCL  with address of sine wave data.
3. Start DMA.
4. Continue with something else.

                         System Response

As soon as DMA starts,

   a. Copy to "back-up" length register from  AUD0LEN .
   b. Copy to "back-up" location register from  AUD0LCL  (will be used
      as a pointer showing current data word to fetch).
   c. Create an  interrupt  for the 680x0 saying that it has completed
      retrieving working copies of  length  and  location registers .
   d. Start retrieving audio data each allocated DMA time slot.

5 / Producing Complex Sounds / Playing Multiple Tones at the Same Time

You can play multiple tones either by using several channels independently
or by summing the samples in several data sets, playing the summed data
sets through a single channel.

Since all four audio channels are independently programmable, each channel
has its own data set; thus a different tone or musical note can be played
on each channel.

5 / Producing Complex Sounds / Modulating Sound

To provide more complex audio effects, you can use one audio channel to
modulate another. This increases the range and type of effects that can be
produced. You can modulate a channel's frequency or  amplitude , or do
both types of modulation on a channel at the same time.

 Amplitude  modulation affects the  volume  of the waveform. It is often
used to produce vibrato or tremolo effects. Frequency modulation affects
the  period  of the waveform. Although the basic waveform itself remains
the same, the pitch is increased or decreased by frequency modulation.

The system uses one channel to modulate another when you attach two
channels. The attach bits in the ADKCON register control how the data from
an audio channel is interpreted (see the table below). Normally, each
channel produces sound when it is enabled. If the "attach"  bit for an
audio channel is set, that channel ceases to produce sound and its data is
used to modulate the sound of the next higher-numbered channel. When a
channel is used as a modulator, the words in its data set are no longer
treated as two individual bytes. Instead, they are used as "modulator"
words. The data words from the modulator channel are written into the
corresponding registers of the modulated channel each time the
 period register  of the modulator channel times out.

To modulate only the  amplitude  of the audio output, you must attach a
channel as a volume modulator. Define the modulator channel's data set as
a series of words, each containing  volume  information in the following

                    Bits      Function
                    ----      --------
                    15 - 7    Not used
                    6 - 0     Volume information, V6 - V0

To modulate only the frequency, you must attach a channel as a period
modulator. Define the modulator channel's data set as a series of words,
each containing  period  information in the following format:

                    Bits      Function
                    ----      --------
                    15 - 0    Period information, P15 - P0

If you want to modulate both  period  and  volume  on the same channel,
you need to attach the channel as both a period and volume modulator. For
instance, if channel 0 is used to modulate both the  period  and frequency
of channel 1, you set two attach bits -- bit 0 to modulate the  volume 
and bit 4 to modulate the  period . When  period  and  volume  are both
modulated, words in the modulator channel's data set are defined
alternately as  volume  and  period  information.

The sample set of data in Table 5-4 shows the differences in
interpretation of data when a channel is used directly for audio, when it
is attached as volume modulator, when it is attached as a period
modulator, and when it is attached as a modulator of both  volume  and
 period .

             Table 5-4: Data Interpretation in Attach Mode

           Independent        Modulating
   Data       (not               Both              Modulating  Modulating
   Words   Modulating)      Period and Volume      Period Only Volume Only
   -----   -----------      -----------------      ----------- -----------
   Word 1  |data|data|  |volume for other channel|  |period|    |volume|
   Word 2  |data|data|  |period for other channel|  |period|    |volume|
   Word 3  |data|data|  |volume for other channel|  |period|    |volume|
   Word 4  |data|data|  |period for other channel|  |period|    |volume|

The lengths of the data sets of the modulator and the modulated channels
are completely independent.

Channels are attached by the system in a predetermined order, as shown in
Table 5-5. To attach a channel as a modulator, you set its attach bit to
1. If you set either the  volume  or  period  attach bits for a channel,
that channel's audio output will be disabled; the channel will be attached
to the next higher channel, as shown in Table 5-5. Because an attached
channel always modulates the next higher numbered channel, you cannot
attach channel 3. Writing a 1 into channel 3's modulate bits only disables
its audio output.

             Table 5-5: Channel Attachment for Modulation

                        ADKCON Register
        Bit    Name     Function
        ---    ----     --------
         7    ATPER3    Use audio channel 3 to modulate nothing
                          (disables audio output of channel 3)
         6    ATPER2    Use audio channel 2 to modulate period
                          of channel 3
         5    ATPER1    Use audio channel 1 to modulate period
                          of channel 2
         4    ATPER0    Use audio channel 0 to modulate period
                          of channel 1
         3    ATVOL3    Use audio channel 3 to modulate nothing
                          (disables audio output of channel 3)
         2    ATVOL2    Use audio channel 2 to modulate volume
                          of channel 3
         1    ATVOL1    Use audio channel 1 to modulate volume
                          of channel 2
         0    ATVOL0    Use audio channel 0 to modulate volume
                          of channel 1

5 Audio Hardware / Producing High-quality Sound

When trying to create high-quality sound, you need to consider the
following factors:

   *  Waveform transitions.
   *  Sampling rate.
   *  Efficiency.
   *  Noise reduction.
   *  Avoidance of aliasing distortion.
   *  Limitations of the low pass filter.

 Making Waveform Transitions      Noise Reduction 
 Sampling Rate                    Aliasing Distortion 
 Efficiency                       Low-Pass Filter 

5 / Producing High-quality Sound / Making Waveform Transitions

To avoid unpleasant sounds when you change from one waveform to another,
you need to make the transitions smooth. You can avoid "clicks" by making
sure the waveforms start and end at approximately the same value. You can
avoid "pops" by starting a waveform only at a zero-crossing point. You can
avoid "thumps" by arranging the average  amplitude  of each wave to be
about the same value. The average  amplitude  is the sum of the bytes in
the waveform divided by the number of bytes in the waveform.

5 / Producing High-quality Sound / Sampling Rate

If you need high precision in your frequency output, you may find that the
frequency you wish to produce is somewhere between two available sampling
rates, but not close enough to either rate for your requirements. In those
cases, you may have to adjust the length of the audio data table in
addition to altering the sampling rate.

For higher frequencies, you may also need to use audio data tables that
contain more than one full cycle of the audio waveform to reproduce the
desired frequency more accurately, as illustrated in Figure 5-4.

 Figure 5-4: Waveform with Multiple Cycles  
Figure 5-4: Waveform with Multiple Cycles

5 / Producing High-quality Sound / Efficiency

A certain amount of overhead is involved in the handling of audio DMA.
If you are trying to produce a smooth continuous audio synthesis, you
should try to avoid as much of the system control overhead as possible.
Basically, the larger the audio buffer you provide to the system, the less
often it will need to  interrupt  to reset the pointers to the top of the
next buffer and, coincidentally, the lower the amount of system
interaction that will be required. If there is only one waveform buffer,
the hardware automatically resets the pointers, so no software overhead is
used for resetting them.

The  Joining Tones  section illustrated how you could join "ends" of tones
together by responding to  interrupts  and changing the values of the
 location registers  to splice tones together. If your system is heavily
loaded, it is possible that the response to the  interrupt  might not
happen in time to assure a smooth audio transition. Therefore, it is
advisable to utilize the longest possible audio table where a smooth
output is required. This takes advantage of the audio DMA capability as
well as minimizing the number of  interrupts  to which the 680x0 must

5 / Producing High-quality Sound / Noise Reduction

To reduce noise levels and produce an accurate sound, try to use the full
range of -128 to 127 when you represent a waveform. This reduces how much
noise (quantization error) will be added to the signal by using more bits
of precision. Quantization noise is caused by the introduction of
round-off error. If you are trying to reproduce a signal, such as a sine
wave, you can represent the  amplitude  of each sample with only so many
digits of accuracy. The difference between the real number and your
approximation is round-off error, or noise.

By doubling the  amplitude , you create half as much noise because the
size of the steps of the wave form stays the same and is therefore a
smaller fraction of the amplitude.

In other words, if you try to represent a waveform using, for example, a
range of only +3 to -3, the size of the error in the output would be
considerably larger than if you use a range of +127 to -128 to represent
the same signal. Proportionally, the digital value used to represent the
waveform  amplitude  will have a lower error. As you increase the number
of possible sample levels, you decrease the relative size of each step
and, therefore, decrease the size of the error.

To produce quiet sounds, continue to define the waveform using the full
range, but adjust the  volume . This maintains the same level of accuracy
(signal-to-noise ratio) for quiet sounds as for loud sounds.

5 / Producing High-quality Sound / Aliasing Distortion

When you use sampling to produce a waveform, a side effect is caused when
the sampling rate "beats" or combines with the frequency you wish to
produce. This produces two additional frequencies, one at the sampling
rate plus the desired frequency and the other at the sampling rate minus
the desired frequency. This phenomenon is called aliasing distortion.

Aliasing distortion is eliminated when the sampling rate exceeds the
output frequency by at least 7 KHz. This puts the beat frequency outside
the range of the  low-pass filter , cutting off the undesirable
frequencies. Figure 5-5 shows a frequency domain plot of the anti-aliasing
 low-pass filter  used in the system.

   0db |________ filter
       |        \ response
       |         \
       |          \
       |           \      filter passes all frequences below about 5kHz.
       |            \
 -30db |_____________\_____ ___________________________________________\
                 |         |         |         |         |         |   /
                5kHz     10kHz     15kHz     20kHz     25kHz     30kHz

           Figure 5-5: Frequency Domain Plot of Low-Pass Filter

Figure 5-6 shows that it is permissible to use a 12 KHz sampling rate to
produce a 4 KHz waveform. Both of the beat frequencies are outside the
range of the filter, as shown in these calculations:

                         12 + 4 = 16 KHz
                         12 - 4 = 8 KHz

      /|\  filter            12kHz sampling frequencey
       |  response             |
   0db |________               |
       |        \     Diff.    |      Sum
       |       | \     |       |       |
       |       |  \    |       |       |
       |       |   \   |       |       |
       |   4kHz|    \  |       |       |
 -30db |_______|_____\_|___ ___|_______|_______________________________\
               | |         |         |         |         |         |   /
                5kHz     10kHz     15kHz     20kHz     25kHz     30kHz
       desired output frequency

          Figure 5-6: Noise-free Output (No Aliasing Distortion)

You can see in Figure 5-7 that is unacceptable to use a 10 KHz sampling
rate to produce a 4 KHz waveform. One of the beat frequencies (10 - 4) is
within the range of the filter, allowing some of that undesirable
frequency to show up in the audio output.

      /|\  filter        10kHz sampling frequencey
       |  response         |
   0db |________           |
       |        \ Diff.    |      Sum
       |       | \ |       |       |
       |       |  \|       |       |
       |       |   \       |       |
       |   4kHz|   |\      |       |
 -30db |_______|___|_\_____|_______|___________________________________\
               | |         |         |         |         |         |   /
                5kHz     10kHz     15kHz     20kHz     25kHz     30kHz
       desired output frequency

                   Figure 5-7: Some Aliasing Distortion

All of this gives rise to the following equation, showing that the
sampling frequency must exceed the output frequency by at least 7 KHz, so
that the beat frequency will be above the cutoff range of the
 anti-aliasing filter :

   Minimum sampling rate = highest frequency component + 7 KHz

The frequency component of the equation is stated as "highest frequency
component" because you may be producing a complex waveform with multiple
frequency elements, rather than a pure sine wave.

5 / Producing High-quality Sound / Low-Pass Filter

The system includes a low-pass filter that eliminates
 aliasing distortion  as described above. This filter becomes active
around 4 KHz and gradually begins to attenuate (cut off) the signal.
Generally, you cannot clearly hear frequencies higher than 7 KHz.
Therefore, you get the most complete frequency response in the frequency
range of 0 - 7 KHz. If you are making frequencies from 0 to 7 KHz, you
should select a sampling rate no less than 14 KHz, which corresponds to a
 sampling period  in the range 124 to 256.

At a  sampling period  around 320, you begin to lose the higher frequency
values between 0 KHz and 7 KHz, as shown in Table 5-6.

          Table 5-6: Sampling Rate and Frequency Relationship

                            Sampling    Sampling    Maximum Output
                             Period    Rate (KHz)   Frequency (KHz)
                            --------   ----------   ---------------
     Maximum sampling rate    124         29              7
     Minimum sampling rate    256         14              7
       for 7 KHz output
     Sampling rate too low    320         11              4
       for 7 KHz output

In A2000's with 2 layer motherboards and later A500 models there is a
control bit that allows the audio output to bypass the low pass filter.
This control bit is the same output bit of the  8520 CIA  that controls
the brightness of the red "power" LED (CIA A $BFE001 - Bit 1: /LED).
Bypassing the filter allows for improved sound in some applications, but
an external filter with an appropriate cutoff frequency may be required.

5 Audio Hardware / Using Direct (Non-DMA) Audio Output

It is possible to create sound by writing audio data one word at a time to
the audio output addresses, instead of setting up a list of audio data in
memory. This method of controlling the output is more processor-intensive
and is therefore not recommended.

To use direct audio output, do not enable the DMA for the audio channel
you wish to use; this changes the timing of the  interrupts . The normal
 interrupt  occurs after a data address has been read; in direct audio
output, the  interrupt  occurs after one data word has been output.

Unlike in the DMA-controlled automatic data output, in direct audio
output, if you do not write a new set of data to the output addresses
before two sampling intervals have elapsed, the audio output will cease
changing. The last value remains as an output of the
 digital-to-analog converter .

The  volume  and  period registers  are set as usual.

5 Audio Hardware / The Equal-tempered Musical Scale

Table 5-7 gives a close approximation of the equal-tempered scale over one
octave when the sample size is 16 bytes.  The " Period " column gives the
period count you enter into the  period register . The length register
 AUDxLEN  should be set to 8  (16 bytes = 8 words). The sample should
represent one cycle of the waveform.

         Table 5-7: Equal-tempered Octave for a 16 Byte Sample

        NTSC    PAL              Ideal   Actual NTSC  Actual PAL
       Period  Period   Note   Frequency  Frequency   Frequency
       ------  ------   ----   --------- -----------  ----------
        254     252      A       880.0      880.8       879.7
        240     238      A#      932.3      932.2       931.4
        226     224      B       987.8      989.9       989.6
        214     212      C      1046.5     1045.4      1045.7
        202     200      C#     1108.7     1107.5      1108.4
        190     189      D      1174.7     1177.5      1172.9
        180     178      D#     1244.5     1242.9      1245.4
        170     168      E      1318.5     1316.0      1319.5
        160     159      F      1396.9     1398.3      1394.2
        151     150      F#     1480.0     1481.6      1477.9
        143     141      G      1568.0     1564.5      1572.2
        135     133      G#     1661.2     1657.2      1666.8

The table above shows the  period values  to use with a 16 byte sample to
make tones in the second octave above middle C.   To generate the tones in
the lower octaves, there are two methods you can use, doubling the period
value or doubling the sample size.

When you double the  period , the time between each sample is doubled so
the sample takes twice as long to play.  This means the frequency of the
tone generated is cut in half which gives you the next lowest octave.
Thus, if you play a C with a  period value  of 214, then playing the same
sample with a  period value  of 428 will play a C in the next lower octave.

Likewise, when you double the sample size, it will take twice as long to
play back the whole sample and the frequency of the tone generated will be
in the next lowest octave.  Thus, if you have an 8 byte sample and a 16
byte sample of the same waveform played at the same speed, the 16 byte
sample will be an octave lower.

A sample for an equal-tempered scale typically represents one full cycle
of a note.  To avoid  aliasing distortion  with these samples you should
use  period values  in the range 124-256 only.   Periods  from 124-256
correspond to playback rates in the range 14-28K samples per second which
makes the most effective use of the Amiga's 7 KHz  cut-off filter  to
prevent noise. To stay within this range you will need a different sample
for each octave.

If you cannot use a different sample for each octave, then you will have
to adjust the  period value  over its full range 124-65536.  This is
easier for the programmer but can produce undesirable high-frequency noise
in the resulting tone.  Read the section called  Aliasing Distortion  for
more about this.

The values in Table 5-7 were generated using the formula shown below.  To
calculate the tone generated with a given sample size and  period  use:

                  Clock Constant         3579545
   Frequency = --------------------- = ----------- = 880.8 Hz
               Sample Bytes * Period   16 * Period

The clock constant in an NTSC system is 3579545 ticks per second.  In a
PAL system, the clock constant is 3546895 ticks per second. Sample bytes
is the number of bytes in one cycle of the waveform sample. (The clock
constant is derived from dividing the system clock value by 2. The value
will vary when using an external system clock, such as a genlock.)

Using the formula above you can generate the values needed for the
even-tempered scale for any arbitrary sample.  Table 5-8 gives a close
approximation of a five octave even tempered-scale using five samples. The
values were derived using the formula above.  Notice that in each octave
 period values  are the same but the sample size is halved.  The samples
listed represent a simple triangular wave form.

               Table 5-8: Five Octave Even-tempered Scale

         NTSC    PAL              Ideal   Actual NTSC  Actual PAL
        Period  Period   Note   Frequency  Frequency   Frequency
        ------  ------   ----   --------- -----------  ----------
         254     252      A       55.00      55.05       54.98
         240     238      A#      58.27      58.26       58.21
         226     224      B       61.73      61.87       61.85
         214     212      C       65.40      65.34       65.35
         202     200      C#      69.29      69.22       69.27
         190     189      D       73.41      73.59       73.30
         180     178      D#      77.78      77.68       77.83
         170     168      E       82.40      82.25       82.47
         160     159      F       87.30      87.39       87.13
         151     150      F#      92.49      92.60       92.36
         143     141      G       98.00      97.78       98.26
         135     133      G#     103.82     103.57      104.17

               Sample size = 256 bytes,  AUDxLEN  = 128

         254     252      A      110.00     110.10      109.96
         240     238      A#     116.54     116.52      116.43
         226     224      B      123.47     123.74      123.70
         214     212      C      130.81     130.68      130.71
         202     200      C#     138.59     138.44      138.55
         190     189      D      146.83     147.18      146.61
         180     178      D#     155.56     155.36      155.67
         170     168      E      164.81     164.50      164.94
         160     159      F      174.61     174.78      174.27
         151     150      F#     184.99     185.20      184.73
         143     141      G      196.00     195.56      196.52
         135     133      G#     207.65     207.15      208.35

               Sample size = 128 bytes,  AUDxLEN  = 64

         254     252      A      220.00     220.20      219.92
         240     238      A#     233.08     233.04      232.86
         226     224      B      246.94     247.48      247.41
         214     212      C      261.63     261.36      261.42
         202     200      C#     277.18     276.88      277.10
         190     189      D      293.66     294.37      293.23
         180     178      D#     311.13     310.72      311.35
         170     168      E      329.63     329.00      329.88
         160     159      F      349.23     349.56      348.55
         151     150      F#     369.99     370.40      369.47
         143     141      G      392.00     391.12      393.05
         135     133      G#     415.30     414.30      416.70

                Sample size = 64 bytes,  AUDxLEN  = 32

         254     252      A      440.0      440.4       439.8
         240     238      A#     466.16     466.09      465.72
         226     224      B      493.88     494.96      494.82
         214     212      C      523.25     522.71      522.83
         202     200      C#     554.37     553.77      554.20
         190     189      D      587.33     588.74      586.46
         180     178      D#     622.25     621.45      622.70
         170     168      E      659.26     658.00      659.76
         160     159      F      698.46     699.13      697.11
         151     150      F#     739.99     740.80      738.94
         143     141      G      783.99     782.24      786.10
         135     133      G#     830.61     828.60      833.39

                Sample size = 32 bytes,  AUDxLEN  = 16

         254     252      A      880.0      880.8       879.7
         240     238      A#     932.3      932.2       931.4
         226     224      B      987.8      989.9       989.6
         214     212      C      1046.5    1045.4      1045.7
         202     200      C#     1108.7    1107.5      1108.4
         190     189      D      1174.7    1177.5      1172.9
         180     178      D#     1244.5    1242.9      1245.4
         170     168      E      1318.5    1316.0      1319.5
         160     159      F      1396.9    1398.3      1394.2
         151     150      F#     1480.0    1481.6      1477.9
         143     141      G      1568.0    1564.5      1572.2
         135     133      G#     1661.2    1657.2      1666.8

                Sample size = 16 bytes,  AUDxLEN  = 8

                            256 Byte Sample

      0   2   4   6   8  10  12  14  16  18  20  22  24  26  28  30
     32  34  36  38  40  42  44  46  48  50  52  54  56  58  60  62
     64  66  68  70  72  74  76  78  80  82  84  86  88  90  92  94
     96  98 100 102 104 106 108 110 112 114 116 118 120 122 124 126
    128 126 124 122 120 118 116 114 112 110 108 106 104 102 100  98
     96  94  92  90  88  86  84  82  80  78  76  74  72  70  68  66
     64  62  60  58  56  54  52  50  48  46  44  42  40  38  36  34
     32  30  28  26  24  22  20  18  16  14  12  10   8   6   4   2
      0  -2  -4  -6  -8 -10 -12 -14 -16 -18 -20 -22 -24 -26 -28 -30
    -32 -34 -36 -38 -40 -42 -44 -46 -48 -50 -52 -54 -56 -58 -60 -62
    -64 -66 -68 -70 -72 -74 -76 -78 -80 -82 -84 -86 -88 -90 -92 -94
    -96 -98-100-102-104-106-108-110-112-114-116-118-120-122-124-126
   -127-126-124-122-120-118-116-114-112-110-108-106-104-102-100 -98
    -96 -94 -92 -90 -88 -86 -84 -82 -80 -78 -76 -74 -72 -70 -68 -66
    -64 -62 -60 -58 -56 -54 -52 -50 -48 -46 -44 -42 -40 -38 -36 -34
    -32 -30 -28 -26 -24 -22 -20 -18 -16 -14 -12 -10  -8  -6  -4  -2

                            128 Byte Sample

      0   4   8  12  16  20  24  28  32  36  40  44  48  52  56  60
     64  68  72  76  80  84  88  92  96 100 104 108 112 116 120 124
    128 124 120 116 112 108 104 100  96  92  88  84  80  76  72  68
     64  60  56  52  48  44  40  36  32  28  24  20  16  12   8   4
      0   4   8  12  16  20  24  28  32  36  40  44  48  52  56  60
     64  68  72  76  80  84  88  92  96 100 104 108 112 116 120 124
   -127-124-120-116-112-108-104-100 -96 -92 -88 -84 -80 -76 -72 -68
    -64 -60 -56 -52 -48 -44 -40 -36 -32 -28 -24 -20 -16 -12  -8  -4

                             64 Byte Sample

      0   8  16  24  32  40  48  56  64  72  80  88  96 104 112 120
    128 120 112 104  96  88  80  72  64  56  48  40  32  24  16   8
      0  -8 -16 -24 -32 -40 -48 -56 -64 -72 -80 -88 -96-104-112-120
   -127-120-112-104 -96 -88 -80 -72 -64 -56 -48 -40 -32 -24 -16  -8

                             32 Byte Sample

      0  16  32  48  64  80  96 112 128 112  96  80  64  48  32  16
      0 -16 -32 -48 -64 -80 -96-112-127-112 -96 -80 -64 -48 -32 -16

                             16 Byte Sample

      0  32  64  96 128  96  64  32   0 -32 -64 -96-127 -96 -64 -32

5 Audio Hardware / Decibel Values for Volume Ranges

Table 5-9 provides the corresponding decibel values for the  volume 
ranges of the Amiga system.

             Table 5-9: Decibel Values and Volume Ranges

            Volume  Decibel Value    Volume  Decibel Value
            ------  -------------    ------  -------------
              64         0.0           32        -6.0
              63        -0.1           31        -6.3
              62        -0.3           30        -6.6
              61        -0.4           29        -6.9
              60        -0.6           28        -7.2
              59        -0.7           27        -7.5
              58        -0.9           26        -7.8
              57        -1.0           25        -8.2
              56        -1.2           24        -8.5
              55        -1.3           23        -8.9
              54        -1.5           22        -9.3
              53        -1.6           21        -9.7
              52        -1.8           20       -10.1
              51        -2.0           19       -10.5
              50        -2.1           18       -11.0
              49        -2.3           17       -11.5
              48        -2.5           16       -12.0
              47        -2.7           15       -12.6
              46        -2.9           14       -13.2
              45        -3.1           13       -13.8
              44        -3.3           12       -14.5
              43        -3.5           11       -15.3
              42        -3.7           10       -16.1
              41        -3.9            9       -17.0
              40        -4.1            8       -18.1
              39        -4.3            7       -19.2
              38        -4.5            6       -20.6
              37        -4.8            5       -22.1
              36        -5.0            4       -24.1
              35        -5.2            3       -26.6
              34        -5.5            2       -30.1
              33        -5.8            1       -36.1
                                        0   Minus infinity

5 Audio Hardware / The Audio State Machine

For an explanation of the various states, refer to Figure 5-8. There is
one audio state machine for each channel. The machine has eight states and
is clocked at the clock constant rate (3.58 MHz NTSC). Three of the states
are basically unused and just transfer back to the idle (000) state. One
of the paths out of the idle state is designed for interrupt-driven
operation (processor provides the data), and the other path is designed
for DMA-driven operation (the "Agnus" special chip provides the data).

In interrupt-driven operation, transfer to the main loop (states 010 and
011) occurs immediately after data is written by the processor. In the 010
state the upper byte is output, and in the 011 state the lower byte is
output. Transitions such as 010->011->010 occur whenever the period
counter counts down to one. The period counter is reloaded at these
transitions. As long as the  interrupt  is cleared by the processor in
time, the machine remains in the main loop. Otherwise, it enters the idle
state.  Interrupts  are generated on every word transition (011->010).

In DMA-driven operation, transition to the 001 state occurs and DMA
requests are sent to Agnus as soon as DMA is turned on. Because of
pipelining in Agnus, the first data word must be thrown away. State 101 is
entered as soon as this word arrives; a request for the next data word has
already gone out. When the data arrives, state 010 is entered and the main
loop continues until the DMA is turned off. The length counter counts down
once with each word that comes in. When it finishes, a DMA restart request
goes to Agnus along with the regular DMA request. This tells Agnus to
reset the pointer to the beginning of the table of data. Also, the length
counter is reloaded and an  interrupt  request goes out soon after the
length counter finishes (counts to one). The request goes out just as the
last word of the waveform starts its output.

DMA requests and restart requests are transferred to Agnus once each
horizontal line, and the data comes back about 14 clock cycles later (the
duration of a clock cycle is 280 ns).

In attach mode, things run a little differently. In attach volume,
requests occur as they do in normal operation (on the 011->010
transition). In attach period, a set of requests occurs on the 010->011
transition. When both attach period and attach volume are high, requests
occur on both transitions.

If the sampling rate is set much higher than the normal maximum sampling
rate (approximately 29 KHz), the two samples in the buffer register will
be repeated. If the  filter  on the Amiga is bypassed and the  volume  is
set to the maximum ($40), this feature can be used to make modulated
carriers up to 1.79 MHz. The modulation is placed in the memory map, with
plus values in the even bytes and minus values in the odd bytes.

The symbols used in the state diagram are explained in the following list.
Upper-case names indicate external signals; lower-case names indicate
local signals.

   AUDxON      DMA on "x" indicates channel number (signal from  DMACON ).

   AUDxIP       Audio interrupt  pending (input to channel from interrupt

   AUDxIR       Audio interrupt  request (output from channel to interrupt

   intreq1     Interrupt request that combines with intreq2 to form

   intreq2     Prepare for interrupt request. Request comes out after the
               next 011->010 transition in normal operation.

   AUDxDAT     Audio data load signal. Loads 16 bits of data to audio

   AUDxDR      Audio DMA request to Agnus for one word of data.

   AUDxDSR     Audio DMA request to Agnus to reset pointer to start of

   dmasen      Restart request enable.

   percntrld   Reload period counter from back-up latch typically written
               by processor with  AUDxPER  (can also be written by attach

   percount    Count period counter down one latch.

   perfin      Period counter finished (value = 1).

   lencntrld   Reload length counter from back-up latch.

   lencount    Count length counter down one notch.

   lenfin      Length counter finished (value = 1).

   volcntrld   Reload volume counter from back-up latch.

   pbufld1     Load output buffer from holding latch written to by AUDxDAT.

   pbufld2     Like pbufld1, but only during 010->011 with attach period.

   AUDxAV      Attach volume. Send data to volume latch of next channel
               instead of to D->A converter.

   AUDxAP      Attach period. Send data to period latch of next channel
               instead of to the D->A converter.

   penhi       Enable the high 8 bits of data to go to the D->A converter.

   napnav      /AUDxAV * /AUDxAP + AUDxAV -- no attach stuff or else attach
               volume. Condition for normal DMA and interrupt requests.

   sq2,1,0     The name of the state flip-flops, MSB to LSB.

 Figure 5-8: Audio State Diagram  
Figure 5-8: Audio State Diagram ECS Audio. ---------- For information on the audio hardware in the Enhanced Chip Set, see the ECS register map in Appendix C.

Converted on 22 Apr 2000 with RexxDoesAmigaGuide2HTML 2.1 by Michael Ranner.