Audio Plus 10-94/AUDIOPLUS.ISO/amiga/samplers/agms/agmsrcrd.doc

Text File | 1991-06-16 | 15KB | 305 lines

AGMSRecordSound is a Modula-2 program which records sound samples to
disk, using a standard parallel port audio digitizer.  AGMSRecordSoundPS3
is a modified version that works with version 3.0+ PerfectSound hardware.

To get you started, try this command:

  AGMSRecordSound ram:test verbose

Type a control-C to stop it.


Propoganda

Since AGMSRecordSound doesn't need to fit the whole sample into one
contiguous memory block, you can record samples longer than your memory
size (up to 2 gigabytes).  It also multi-tasks so you can see the screen
and do other things while recording, unlike all other sound digitizer
programs I know of.

Anti-propoganda

The downside of this is crummy audio quality.  Other digitizers take
over the computer while recording since that is the best way to get
better quality.  The trouble comes from code that temporarily disables
interrupts, making AGMSRecordSound's interrupt routine miss a sound
sample.  This happens all time time - whenever memory is allocated or
deallocated, interrupts are disabled to prevent other programs from
interfering.  Hopefully Commodore will put in some DMA circuits for
digitizing so that we can get good quality and multitasking.

Requirements

AGMSRecordSound requires sound sampler hardware plugged into the
parallel port.  I've tested it with AMAS (records from the right
channel) and it should work with most other parallel port digitizers
(try left if right doesn't work).  If you have SunRize Industries'
PerfectSound 3 digitizer then you have to use a different version of
this program: AGMSPlaySoundPS3.  You also need to have the arp.library
file (version 39 or 1.3) in your LIBS: directory.  Other than that, it
should work on all Amigas.

How It Works

An interrupt routine is set up for audio channel 1 (right side) that
samples the parallel port, copies the data to the audio register and
copies it to a buffer in memory.  Every time audio channel 1 needs a new
sound sample, this interrupt routine is called.  When a buffer becomes
full, it moves to the next buffer and signals the main program.  If
there are no empty buffers to be filled, the interrupt routine will slow
down the audio channel to a lower sampling rate (around 440 Hz) so that
the processor will have more CPU time for running tasks (less used by
the interrupt code).  You can still hear the sounds but they won't be
stored in the buffer (not recorded in other words) and they will sound
strange (like a dog growling under water) due to the lower sampling
rate.  When an empty buffer becomes available, it will resume filling at
normal speed.

The other half of AGMSRecordSound is a process that dumps filled buffers
to a disk file.  If it is faster at emptying buffers than the interrupt
routine is at filling them, it will go to sleep and wait for a signal:
control-C (a signal from the user to stop recording) and control-F (a
signal from the interrupt routine that another buffer is full).  In the
PerfectSound version, it also responds to control-E (steps up the volume
control).

Sound Quality

The problems with disabled interrupts cause pops and similar noises -
great if you are trying to simulate an old scratchy record (or to record
one), but otherwise annoying.  The trick for getting "good" sound
quality is to set the sampling rate to minimize this problem.

A really low sampling rate means that the disabled interrupt won't last
long enough to cause trouble.  After interrupts are re-enabled, the
pending audio interrupt will get through and the sample will only be
delayed in time for a short while.  Samples after the delayed one will
be back on schedule.  Unfortunately, most people don't like to record at
1kHz :-).  However, if you have a faster computer (I only have an A2000)
then the disabled times will be shorter and sound quality should go up.

A really high sampling rate will suck up so much CPU time that only the
interrupt routine will be running; every time it finishes sampling,
another interrupt comes in.  This is actually quite useful (more later).
You can tell that this is happening by the lack of mouse pointer
movement when you move the mouse.  However, as soon as all the buffers
are full, the interrupt routine will go into idle mode.  This frees up
CPU cycles for running other tasks - like the mouse pointer movement and
buffer saving.  As soon as one buffer has been saved, the interrupt
routine will pick up speed and things will be frozen.

At high sampling rates, no other task is running and disabling
interrupts.  This means high quality sound, almost as good as those
dedicated non-multitasking sampler programs.  The catch is that nothing
is recorded while the buffer is being dumped to disk.  So, if you can
put up with brief pauses then you can use this mode.

Also, you can use AGMSRecordSound as an ordinary sound digitizer.  Just
set the buffer size to fill up your whole memory (many smaller buffers
are preferable since the interrupt routine can fill all but the one
buffer being emptied - try buffers 20 size 100000 if you want to fill up
2 meg of memory).  Then set the sampling rate to one that uses all the
CPU time for sampling (12kHz is good on my A2000, you will need larger
rates on faster computers).

At ridiculously high sampling rates, the interrupt routine will be
running flat out at a speed less than the one you requested.  This will
result in a sample that is higher pitched than reality when you play it
back.

IFF Information

The output file is an IFF file.  You have lots of command line options
available for specifying optional information (like sample name, author
etc).  Unfortunately some other sound sample processing programs can't
handle the extra info (but AGMSPlaySound can!).  If you find that you
want to type in more information than will fit on a command line, put
the long command line into a script file and execute it.

Help

Most of the arguments to the program are obvious if you know audio
terminology and computer terminology.  Anyways, you really don't have to
specify any arguments except the name of the file to be created.  For
on-line help, just type a ? as the argument to get the standard ARP /
AmigaDOS command template.  A second ? typed at the template prompt will
print the program credits.  The Verbose option also explains in more
detail what the current settings are.  They are intuitively obvious
(well, to me :-), but just in case you aren't intuitive about audio
stuff, I'll describe the parameters in detail near the end of this
message.

Stolen Channels

This program won't let other programs steal its sound channel (channel 1
to be specific).  It allocates it with maximum priority and thus hangs
on to it.  If you are using AGMSPlaySound at the same time as you are
recording, AGMSPlaySound will be bumped off channel 1 and will reappear
on whatever other channel is free.  If someone else has channel 1 with
the highest priority then AGMSRecordSound won't try recording.

Error Handling

The main source of errors is from the file system.  In particular, you
may encounter a disk full error once in a while :-).  When an error
occurs, AGMSRecordSound stops whatever it is doing and shuts down.  It
will return a return code of 10 when something goes wrong.

Now, more about those disk full errors.  I've set things up so that the
file you are recording to is closed as if you had normally ended
recording.  Since it is an IFF file, this means going back and filling
in the size of the sample in previously written data near the beginning
of the file.  It seems to work when using the old file system (OFS) but
doesn't work under the fast file system (FFS).  Well, I tried.

Technical Notes

Well, I found out a bit while writing this program.  Ok, two bits, make
that three or four after I figured out the PS3 hardware :-).  For one
thing, I found out how to use a digitizer: set CIAA port B to be all
inputs (the parallel port).  Set CIAB port A pins POUT and SEL to be
outputs.  The sampler spews out bytes to the parallel port with no
handshaking, at some rate that I assume is pretty fast.  Some digitizers
use a high level on SEL to connect the right audio input to the sampler
and a high level on POUT to connect the left channel.  AMAS just
seems to use the POUT bit to select left or right and seems to use right
instead of left.  Whatever.  Once you have the desired channel, you can
get a sample byte by reading the parallel port byte.  Subtract $80 from
it to make it into a signed number and store it away.  Allow a small
amount of time between switching from left to right (or vice versa) to
let the sampler update its value (the time for a few instructions should
be enough).

The PerfectSound 3.0+ hardware is a different story.  The sample byte is
split between the parallel port (CIAA port B) and the miscellaneous bits
port (CIAB port A).  The low 6 bits of parallel data (PB5-PB0 in CIAA)
are actually the high six bits of the sample (shift them left by two).
The low two bits of the sample are in the printer POUT (paper out) and
BUSY signals (PA1-PA0 in CIAB).  As before, SEL (PA2 in CIAB) controls
the left / right (zero is left).  So, what do PB6 and PB7 in CIAA do? A
lot! The PerfectSound digitizer has a latched output, PB6 triggers the
latch.  Set PB6 to zero to latch a value, read the sample, and set PB6
back to one to start digitizing the next sample (I assume that's what
it's doing).  PB7 is the gain control.  Normally you run the hardware
with PB7 set to one.  If you toggle PB7 to zero and back, the volume
control will be stepped up or down one notch of 16 levels.  SEL controls
the volume direction (as well as left/right), zero for decreasing the
volume.  The volume gain control wraps around from minimum to maximum
and as far as I could tell, there isn't any way of telling what the
current gain setting is or of setting it to minimum or maximum.

I experimented a bit with ways of generating interrupts before I clued
in that no interrupts happen while interrupts are disabled.  CIA timers
running continously are good.  Audio channels themselves are also good
for making a periodic interrupt (and at exactly the same speed as
playback too).  I even tried using an audio channel to generate an
interrupt slightly before the audio channel I was using for timing would
signal the next time period.  The early interrupt routine would then
busy wait for the timing channel interrupt.  The "slightly" grew to over
800 CPU clock cycles before I got slightly improved sound (hey - a way
of determining how long those disabled times are), slightly improved at
a very slow sampling rate that made it useless.

Feedback

If you have any feature requests or find any bugs, please send me a
message.  I'm on several of the Ottawa BBX's (a BBS written by SteveX)
and my commercial info service names are listed in the program's second
help message.

File/A

Names the file where the sample will be stored.  Any existing file by
this name will be erased.  A required keyword.

Verbose/S

When this command switch is used, lots of interesting messages will be
displayed.  All the parameter settings will be shown.  During recording
you can see what buffers are being dumped.

Buffers/K

This parameter specifies the number of buffers (any old ram is used) to
use for recording the sound.  You need at least 2.  While the system is
recording one buffer, the others can be emptied to disk.  Try the
verbose command switch to watch the buffers being emptied and to see how
changing the number of buffers affects performance.  Gee, I sure like
these reuseable documentation paragraphs - I just changed "play" to
"record" :-)

Size/K

This parameter sets the size of each sound buffer.  The default is
100000 bytes, the maximum is about 2000000000 bytes :-).  As mentioned
elsewhere, AGMSRecordSound doesn't need to use any chip ram at all.
Buffers use general purpose ram.  For best performance, use large
buffers.  That is because DOS can write a large chunk of data faster
than several small chunks (less overhead).

Hz/K

This parameter controls the record rate.  It is in units of samples per
second, or bytes per second since each sample is one byte long (compact
disks have 2 byte samples and thus sound better).  The default is 7781
hz, less for the PerfectSound version due to extra overhead in reading
each sample.  The highest rate on my A2000 that lets you record and
simultaneously save to disk (no gaps) is about 8500 hz.  A speed of
12000 hz is good for better quality sound with small gaps between
buffers.  See the discussion on sound quality about picking values for
hz.

The Hz setting is internally translated into a code value for the
hardware.  Since the hardware doesn't have all that many code values,
only a few frequencies are actually available.  A frequency near the one
you specified will be picked.  Note that this is half the playback code
value since playback plays words (pairs of bytes) while we only record a
byte at a time.

Name/K

This lets you specify the name of the song or whatever you have
recorded.  If it is more than one word, put it in double quotes.

Copyright/K

This is for the copyright notice.  Specify something like "1991,
Alexander G. M. Smith".

Author/K

The author of this sample.  If it is music then that is relatively
obvious.  I don't know who is the author of the sound of a toilet
flushing, maybe the toilet manufacturer? Don't specify it if you don't
want an author.

Anno=Annotation/...

An endless list of annotation.  You can use the short form of the
keyword (Anno) or the long form (Annotation).  The values are strings
(as many as you want) of annotation text.  It can be whatever you want,
usually something that describes the sample: anno "This is my first hour
long sound sample" "The sound of the dripping tap was created by using
the kitchen sink." "Use at your own risk."

Control-C Etc.

Control-C will abort the program as soon as possible.  This can take a
while if it is busy dumping a buffer.

Control-E will step up the volume on the PerfectSound hardware, if you
are using AGMSRecordSoundPS3 (does nothing otherwise).  Don't hit the
key too fast, when AGMSRecordSoundPS3 is running the rest of the
operating system is slowed down and can't process your signals fast
enough.  In fact, it's so slow that you can overflow something (probably
the supervisor stack) by typing a lot of Control-CDEF's, even in another
program's window.  The result is a visit from the guru (crash #3
usually).

Control-F is used by the interrupt routine to tell the main program that
there is another full buffer.  You can use it to if you want to wake up
a poor little sleepy program (doesn't do any harm, unless you type it
too quickly :-).

- Alex