Amiga Collections: Various/DevDisk 65 (1989)(DevWare PD).zip/DevDisk 65 (1989)(DevWare PD).adf/scales/scales.doc

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Article 182 of net.micro.amiga:
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From: [e-mail] (John Draper)
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Subject: Sound Tutorial by Steven A. Bennett from BIX
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Date: 11 Jan 86 11:23:49 GMT
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A Discussion of Simple Audio Generation on the Amiga
= ========== == ====== ===== ========== == === =====
by Steven A. Bennett

(permission is given to freely distribute this discussion and
its associated source file, SCALES.C, as long as due credit is given)


Introduction
------------

  Audio generation on the Amiga is not the easiest of tasks, but once
one has the basic knowledge involved, it becomes no more than a matter
of writing a good set of interface routines.  The Amiga can generate
four channel sound through use of four DMA-driven Digital to Analog
converters supplied on one of the custom chips. (Portia, I believe.)
These channels can be used to generate music, speech, and sound effects
with little processor overhead, in most cases.

  In this discussion I will focus mainly on simple tone generation
using several channels.  Also, I have uploaded a C source file called
SCALES.C to the Listings conference here on BIX, and will be referencing
sectons of code in it during the discussion.  It is suggested that you
download this file as an example of the interface described below.

  One further note -- My knowledge of the Audio device is based on the
small section of the 1.0 ROM Manual which my dealer graciously allowed
me to copy.  Thus, I may turn out to be incorrect on some of the non-Audio
related areas of this discussion.  Since these are few, however, I feel able
to upload this with little worry.


The Audio Device
--- ----- ------

  In the back of the ROM Kernal Manual is a description of the subroutines
available in the Audio Device.  Most of the information, while perhaps
useful for some complicated tasks, is unnecessary for purposes of this
discussion.  The routines which are needed here are listed below:

               BeginIO()      - Begin an audio IO task
                  CMD_WRITE      - Set waveform and begin audio output
                  ADCMD_PERVOL   - Change period and volume
                  ADCMD_FINISH   - Finish the current CMD_WRITE
               OpenDevice()   - Open the audio device
               WaitIO()       - Wait for the CMD_WRITE to finish

The commands listed under BeginIO() are all that are needed for simple
Audio output.  Required for all of these commands is the control structure,
struct IOAudio.  It is defined in devices/audio.h.


Preparing the Driver
--------- --- ------

  In order to use the Driver, one needs an IOAudio structure.  This should
be public, and all unused fields zeroed out, hence we will allocate it
using AllocMem():

      struct IOAudio *ioa;

      ioa = (struct IOAudio *)AllocMem(sizeof(*ioa) * NBR_IOA_STRUCTS,
         MEMF_PUBLIC | MEMF_CLEAR);

  NBR_IOA_STRUCTS is the total number of IOAudio structures used by this
program.  Since most programs will need more than one, we can allocate them
all at once.  As always, when performing an allocation, the return ought to
be checked for an error.

  Now we must open the Audio Device.  One of the items necessary for this
process is a ReplyPort.  Later on, we will need several more.  (NOTE - this
area is the shakiest one in this program, as I have determined the use of
CreatePort from other programs and lots of guesswork)

      ioa->ioa_Request.io_Message.mn_ReplyPort =
         CreatePort("Audio one", 0);

  After we have the ReplyPort, we have to set the other variables for the
OpenDevice.  Here, we wish to allocate the channels (which can also be done
later, using ADCMD_ALLOCATE, in which case we don't even need the ReplyPort,
but it is far easier to do it here.) which will be used for output.  For
this process, we need an allocation maps.  This is an array of bytes, each
of which contains a bit map for the channels to be requested.  The four
channels are represented by the lowest four bits in each byte.  Thus, if
we wanted a map which would allocate any combination of two channels, our
map might look like this:

      UBYTE aMap[] = { 0x03, 0x05, 0x09, 0x06, 0x0A, 0x0C };

which is all the combinations of two bits in the lowest four.  But for this
example we simply want all four, so our map looks like this:

      UBYTE aMap[] = { 0x0f };

Now we can open the Audio Device:

      /* ln_Pri is the priority of the allocation request.
       * if channels are in use, and ln_Pri is higher than the
       * priority of the channel already in use, it will steal
       * the channel(s) if it can't otherwise allocate them
       * legally.  See ADCMD_ALLOCATE for more information on this.
       */
      ioa->ioa_Request.io_Message.mn_Node.ln_Pri = 10;
      ioa->ioa_Data = aMap;
      ioa->ioa_Length = sizeof(aMap);
      error = OpenDevice(AUDIONAME, 0, ioa, 0);

If there was no error, then ioa->ioa_Request.io_Unit will be a bitmap
of the successfully allocated channels.  When allocating less than four
channels, one might need to look at this value to see exactly which
channels it is legal to use.  An observant C programmer might notice that
io_Unit is a pointer (as does the C compiler, itself).  Nevertheless, it
is actually used as a bitmap, and warnings generated by the compiler on
this point should be ignored.

  Now, given the opened device, one must initialize the rest of the IOAudio
structures to be used by the program.  Since structure assignment in indeed
implemented in the Lattice compiler (and probably the Aztec, but I can't be
certain), this is easily accomplished.  The reason we must set each struct
equal to the IOAudio struct returned by the OpenDevice is to copy the
Device code and allocation code to the new structure.  If this is not done,
then any command using the uninitialized structure will fail.

      finishioa = &ioa[1];  /* Remember the AllocMem()? */
      *finishioa = *ioa;

  One absolutely must have one IOAudio structure per voice which will be
playing simultaneously.  This is because the CMD_WRITE command is
asyncronous, and once it has been started, the IOAudio structure is
supposed to be left untouched.  For purposes of quick change, I am using
two structures per voice.  All of these structures must have their own
ReplyPorts, as well.

  One also needs at least one IOAudio structure for the synchronous
commands ADCMD_PERVOL and ADCMD_FINISH.  I am using seperate structures
for these two, again for speed.

  To see an example of initialization of these structures, examine
the routine InitIOA() in SCALES.C.


Generating a tone
---------- - ----

  In order to generate a tone, one needs several things.  A waveform array,
usually in a length which is a power of two.  A period, which is calculated
from the frequency of the tone and the length of the waveform array.  A
volume and a duration.  And an initialized IOAudio structure.

  First, we'll determine the period.  This must be an integer between
127 and 65535, although best results occur between 127 and 500, due to
the anti-aliasing filter.  (At 500, the sampling rate is only about
7150 times per second, and that is getting slow.  I could even conceive
of an overtone being produced if one gets much higher.)  The period is
taken from the formula:

            P = C/(WF)

where P is the period, C is the clock rate (in this case, 3579545), W is
the length of the waveform array, and F is the frequency in hz.  Thus for
a middle A (440 hz) with, say, a 32 byte long waveform array, the period
would be:

            P = 3579545 / (32 * 440)
              = 3579545 / 14080
              = 254.229

Since we require an integer, this would become 254.  For those of you who
haven't yet burned your ABASIC manuals, there is a table of period values
on page R-138.  For the sample program, we are using period values between
428 and 226, so that our octaves start with C.  This has the advantage of
lowering frequency error to about .22%, but the disadvantage of being
unable to hit the highest notes.  If one can accept a frequency error of
up to .39%, then the range of 240 to 127 could be used.  (For those of
you wondering what those percentages come out to, its about a twentieth
of a step for the former, and a tenth of a step for the latter.)

But how can we change octaves?  By doubling or halving the length of the
waveform array.  So now we turn to waveforms.

A waveform array describes one cycle of a tone.  It can be simple (ie.,
BYTE wfarray[2] = {127, -127}; which is a square wave) or it can be
complex, involving complex transformations using harmonics.  In our
sample program, we use a simple sawtooth wave, which is simply a ramp
from the lowest value to the highest, then suddenly dropping to the lowest
and beginning again.  The length of the waveform array determines the
octave.  Since middle A (fourth octave) uses a waveform array length of
32, we can get the following size table for the various octaves:

               Octave 0 (lowest) = 512
                        Octave 1 = 256
                        Octave 2 = 128
                        Octave 3 = 64
                        Octave 4 = 32
                        Octave 5 = 16
                        Octave 6 = 8
                        Octave 7 = 4
                        Octave 8 = 2
                        Octave 9 = 1    (this is impossible)

Because a sawtooth is a naturally rough wave, Octave 0 would sound pretty
badly, therefore we will not use it for now.  Octave 9 is impossible for
the period range we have chosen, although some of that octave could be
played if we used the lower period range mentioned earlier.  It is
impossible because we need at a minimum 2 samples to make a waveform.
Octave 8 is currently ignored for this program because the expansion
algorithm used would not produce an acceptable 2 byte waveform.

  Therefore, we create a basic, 256 byte, sawtooth wave.  The routine
setwave() in SCALES.C performs this function well.  It is better
if the wave can begin and end at the zero point, allowing for less chance
of click when a note is changed.  A flag (ADIOF_SYNCCYCLE) can be used with
the ADCMD_FINISH command to end the CMD_WRITE in progress at the end of the
currently playing cycle to aid in this, but is unnecessary for our example.

  We must then expand that waveform to several waveforms of decreasing
length, as set in the table of lengths above.  This is accomplished with
the routine xpandwave().  The routine makewaves() will allocate memory for
the wave and call both.  And here we come to an important point.

  The waveform array MUST be in chip memory!  Either it can be kept there
at all times, or copied there just before use.  Do NOT assume that your
program will reside in chip memory and ignore this requirement, for sooner
or later you will be burnt!  Furthermore, I will make an educated guess that
the waveform array should not be modified while in use.  I suspect that the
CMD_WRITE command simply sets registers in the chip (at least, I hope that
is what it does!) and that modifying the waveform while in use may cause
unusual results.

  Finally, now that we have our period and our waveforms, we can make a
tone.  To do this we need to set several entries in a currently unused
IOAudio structure:

      freeioa->ioa_Request.io_Command = CMD_WRITE;
      freeioa->ioa_Request.io_Flags = ADIOF_PERVOL | IOF_QUICK;
      freeioa->ioa_Request.io_Unit = unitno;

      freeioa->ioa_Period = period;
      freeioa->ioa_Volume = 32;            /* between 0 and 64 */
      freeioa->ioa_Data = waveform_ptr;
      freeioa->ioa_Length = waveform_len;

      freeioa->ioa_Cycles = 0;

This assumes that we have already copied the OpenDevice info into it and
have created a unique ReplyPort for it.  In our sample program, this would
be an array.  CMD_WRITE is the command to set waveform and begin audio
generation.  The flag ADIOD_PERVOL says to set period and volume as well.
The flag IOF_QUICK is a flag telling to perform this as quickly as
possible.  It is used here to make CMD_WRITE synchronous in case of an
error.  The variable unitno is a bitmap for the channel(s) for which
to play the tone.  (1 for channel 1, 2 for channel 2, 4 for channel 3, and
8 for channel 4, or any combination thereof.)  ioa_Cycles is the number of
cycles which the tone will play for.  If set to zero, the tone will play
continuously until aborted by AbortIO() or by an ADCMD_FINISH command.  The
latter is recommended.

  Since we set ioa_Cycles to zero, we must consider the possibility that
the audio channel we wish to use is already playing a tone.  A flag should
be kept concerning this fact.  If a tone is already playing, we must stop
it first, which requires an ADCMD_FINISH command to be issued.  The
finishioa structure does not need a ReplyPort, as ADCMD_FINISH is
synchronous.  This initialization, with the exception of the unitno,
can be done once, with our main initialization, but is shown here for
clarity.

   if (waiting[voice])
      {
      finishioa->ioa_Request.io_Command = ADCMD_FINISH;
      finishioa->ioa_Request.io_Flags = IOF_QUICK;
      finishioa->ioa_Request.io_Unit = unitno;
      BeginIO(finishioa);
      WaitIO(ioainuse);
      waiting[voice] = NO;
      }

ioainuse is the IOAudio structure which was used for the previous
CMD_WRITE on this channel.  After it has been told to finish, it sends a
message that it has done so to the ReplyPort specified in it.  WaitIO()
waits until this message has been received, and then continues.

  Finally we can start our tone.  This is simply done, by:

   BeginIO(freeioa);
   error = CheckIO(freeioa);
   if (error)
      <do error recovery>
   else
      waiting[voice] = YES;

In our sample program, we then swap the freeioa and ioainuse pointers, so
that we can easily use the same code for the next tone.  All of this code
is in setwpv() in SCALES.C for your perusal.

All that remains is a routine which will select the waveform to be used and
the period, and this can be a table lookup.  To make things easier on the
driver, however, we can make a simple routine which will just change the
period if the waveform does not change.  This routine, which can also
control the volume, can be used whenever the octave is the same, and can
come in handy later should we desire envelope control or vibrato:

   ioapv->ioa_Request.io_Command = ADIOF_PERVOL;
   ioapv->ioa_Request.io_Flags = IOF_QUICK;
   ioapv->ioa_Request.io_Unit = unitno;

   ioapv->ioa_Period = period;
   ioapv->ioa_Volume = 32;
   BeginIO(ioapv);

Since ADCMD_PERVOL is synchronous, it requires no wait and no ReplyPort.
This should only be used if a CMD_WRITE on the given unitno is in progress.
If not, I have no idea as to the results.

  The waveform and period select routine, which also decides if it can
use the ADCMD_PERVOL code, is called strike() in SCALES.C.  It also
handles "rests" by shutting the volume down to zero.  For determining
durations of notes, one can use the Delay() function, although for
serious music generation, I would suggest the microhz timer instead.


Cleaning up
-------- --

  When your program ends, it is always important to free up the allocated
memory, the ReplyPorts, and the device.  For simplicity, I am using the
FinishProg() routine in SCALES.C.  If you don't do it, nobody will.

  If you are still confused, examine SCALES.C, where all this stuff is
used in a working application.  All subroutines in there can be used by
anyone who wants to.

Other applications
----- ------------

  There are many other uses for the audio routines.  Digitized sound could
be used in place of a waveform for interesting effects, although the period
would most likely require change.  Sound effects could be created by quickly
changing period and waveform and volume.  Supposedly, one can link pairs of
channels for modulation, but I have no idea as to how to do it.  And speech
can be generated, but not with these routines.  Have fun!

S. A. Bennett
12/31/85
===End of Discussion===