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OK, here is the documentation for STE DMA sound output. More documentation
when I've typed it in. Screen blanker postings when I've got a copy of uuencode
... please be patient


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Information from "Atari STE Developer Information Addendum"

STE DMA Sound registers

Register    Access      Description
FF8900      R/W      00 - sound disabled (reset state)
                     01 - sound enabled, disable at end of frame
                     11 - sound enabled, repeat forever

FF8902      R/W      Frame base address (high)
FF8904      R/W      Frame base address (middle)
FF8906      R/W      Frame base address (low)

FF8908      RO       Frame address counter (high)
FF890A      RO       Frame address counter (middle)
FF890C      RO       Frame address counter (low)

FF890E      R/W      Frame end address (high)
FF8910      R/W      Frame end address (middle)
FF8912      R/W      Frame end address (low)

FF8920      R/W      Sound mode control: 
                     xxxx xxxx m000 00rr
                     Where for m:
                       0 Stereo mode
                       1 Monophonic mode
                     Where for rr:
                       00  6258 Hz sample rate (reset state)
                       01 12517 Hz sample rate
                       10 25033 Hz sample rate
                       11 50066 Hz sample rate

FF8922      R/W      MICROWIRE data register
FF8924      R/W      MICROWIRE mask register

Volume/controller commands (device address is always 10)
--------------------------------------------------------

011 DDD DDD Set master volume
    000 000 -80 dB
    010 100 -40 dB
    101 XXX   0 dB

101 xDD DDD Set left channel volume
     00 000 -40 dB
     01 010 -20 dB
     10 1xx   0 dB

100 xDD DDD Set right channel volume
     00 000 -40 dB
     01 010 -20 dB
     10 1xx   0 dB

010 xxD DDD Set treble
      0 000 -12 dB
      0 110   0 dB
      1 100 +12 dB

001 xxD DDD Set bass
      0 000 -12 dB
      0 110   0 dB
      1 100 +12 dB

000 xxx xDD Set mix
         00 -12 dB
         01 Mix GI sound output (ST sound chip)
         10 Do not mix GI sound output
         11 Reserved
----------------------------------------------------------


Sampled sound data is stored in memory as a series of bytes, which 
represent a speaker displacement from -128 to +127. Zero represents the 
neutral or middle speaker position. Playback is programmable at one of 
four rates : 50kHz, 25kHz, 12.5kHz or 6.25kHz.

During the horizontal blanking phase, samples are fetched from memory by 
the DMA sound chip, and fed into a Digital to Analogue Converter (DAC). 
The output of the DAC is then filtered by a four-pole low pass filter to 
a frequency equal to around 40% of the sample frequency. The signal then 
passes through a two pole 16kHz low-pass filter, and fed into the 
National Semiconductor Volume/Tone controller (LMC1992). The final 
output is available from the RCA jacks on the back of the STE, which can 
be fed into an amplifier and hence to speakers, headphones etc.

Both stereo and mono sample replay is provided, but both stereo channels 
are mixed along with the ST's sound chip output for monitor speaker 
output. Sound chip output can also be sent to the stereo output jacks as 
well.

In stereo playback mode, the same data is regarded as words, with the 
high byte of the word being the sample for the left channel, and the low 
byte the right channel sample. In mono mode, each byte is output to both 
left and right stereo channels, but data is still fetched one word at a 
time. This means that mono sample data must always be an even number of 
bytes.

Samples are grouped together in frames. Each frame can be played once, 
or repeated automatically forever (until stopped). Two registers are 
loaded with the frame start and end address - the end address is 
actually the first byte beyond the end of the sample. Thus a 512 byte 
sample with a frame start address of 101024 would have a frame end 
address of 101536. Table One gives the location and description of each 
DMA sound register.

Actually playing a sample is really quite straightforward. Simply 
assemble the data in memory, load the start and end addresses, set 
stereo or mono mode and the playback frequency. Finally, write a one to 
the sound control register, and the sample will play once.

Producing continuous sound and linking frames together are the next 
steps, and hardware support is provided for these processes. The DMA 
sound chip produces a 'DMA sound active' signal which is connected to 
the external input of MFP Timer A. This signal is a one when samples are 
being played, and zero otherwise. At the end of a repeated frame, this 
line goes from one to zero, and then back to one again. Thus setting 
Timer A into event countdown mode allows you to generate an interrupt 
when a frame has been played a set number of times.

Frame repetition is seamless - there is no time delay between the end of 
a sample, and the start of it's replay, because the frame start and end 
registers are double buffered. Writing to these registers actually 
places the data into a holding area, and the contents of the holding 
area actually go into the true registers when the chip is idle (at the 
end of the frame, if one is currently being played).

Thus, if you wanted to play two consecutive frames, you would write the 
start and end addresses, and set the control register to three. The 
first frame will start playing, and you can immediately write the start 
and end addresses of the next frame, without waiting for the first frame 
to finish. There will still be an interrupt from Timer A at the end of 
the first frame, and you could use that to load the address of a further 
frame, and so on.

One further thing to note is that the 'DMA sound active' signal is also 
exclusive-ORed with the 'monochrome monitor detect' signal, and fed into 
the GPIP I7 input of the MFP. This was provided to enable interrupt 
driven sound without using the last free timer of the MFP. It is a 
little more difficult to use, since you will get a different signal edge 
depending on whether a mono or colour monitor is attached, as well as an 
interrupt at the end of every frame.

Monochrome monitors ground the 'mono detect' line, resulting in a zero 
when the bit is read from the MFP. Colour monitors don't ground the line 
(it is left floating), and the bit reads one. When DMA sound is active, 
this situation is inverted (because of the XOR with the 'DMA sound 
active line'). TOS actually looks at this bit during vertical blank 
time, to see if the monitor has been changed, but TOS on any machine 
with the DMA sound chip has been appropriately modified to avoid 
problems.

Finally, the 'DMA sound active' line goes from active to idle (one to 
zero) after the last sample has been fetched. There is a four-word FIFO 
(First In, First Out) buffer inside the chip, so it will be eight sample 
times (in stereo mode) before the sound actually finishes. If you do not 
reload the frame registers in this time, then the join between samples 
will not be seamless.


The volume and tone controller of the STE is connected via a MICROWIRE 
bus interface. The idea behind this is that further devices can be added 
to the bus in the future. The MICROWIRE bus is a simple three wire 
serial connection, with a protocol to allow multiple devices to be 
controlled individually.

In the general case, the data stream consists of N address bits, 
followed by zero or more don't care bits, and then M bits of data. The 
actual hardware interface in the STE consists of two 16 bit read/write 
registers, one for the data to be shifted out, and a mask indicating 
which bits are valid.

A one in any bit of the mask indicates that the corresponding bit in the 
data register is valid. Data transmission starts as soon as the data 
register has been written to, so the mask register must be loaded first. 
Sending takes approximately sixteen micro-seconds, and if the data 
register is read during this time, a 'snap-shot' of the data being 
shifted out will be obtained. This means that if you wait for either 
register to return to its original state, you can be sure that sending 
has been completed.

The volume/tone controller is addressed by a two bit address field of 
%10 (binary) and a nine bit data field. Table One details the commands 
that can be sent to the device, and the addresses of the MICROWIRE 
registers in the STEs memory map. Actually sending these commands is 
easier than it looks. Simply set the mask register to $07FF, and place 
the data in the lower nine bits with %10 in the upper two bits.

For example, setting the mask to $07FF and the data register to $04C4 
will set the master volume to $14. That's all there is to it!


Regards,

Mathew Lodge
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