[AD] play_audio_stream doc patch

[Robert Ohannessian <ohannessian@xxxxxxxxxx>]

If there are no objections, I'll commit this patch.


--
- Robert Jr Ohannessian
http://bob.allegronetwork.com/
The peer will come and reset your connection. RUN WHILE YOU STILL CAN!

Index: docs/src/allegro._tx
===================================================================
RCS file: /cvsroot/alleg/allegro/docs/src/allegro._tx,v
retrieving revision 1.112
diff -u -r1.112 allegro._tx
--- docs/src/allegro._tx	9 Dec 2002 16:40:32 -0000	1.112
+++ docs/src/allegro._tx	12 Dec 2002 02:30:44 -0000
@@ -5853,15 +5853,18 @@
 @xref install_sound, get_audio_stream_buffer, stop_audio_stream
    This function creates a new audio stream and starts it playing. The 
    length is the size of each transfer buffer (in samples), which should 
-   normally be a power of two somewhere around 1k in size: larger buffers 
-   are more efficient and require fewer updates, but result in more latency 
-   between you providing the data and it actually being played. The bits 
-   parameter must be 8 or 16, freq is the sample rate of the data, and the 
-   vol and pan values use the same 0-255 ranges as the regular sample 
-   playing functions. If you want to adjust the pitch, volume, or panning of 
-   a stream once it is playing, you can use the regular voice_*() functions 
-   with stream-&gtvoice as a parameter. The sample data is always in unsigned
-   format, with stereo waveforms consisting of alternate left/right samples.
+   normally (but doesn't have to) be a power of two somewhere around 1k in
+   size. Larger buffers are more efficient and require fewer updates, but
+   result in more latency between you providing the data and it actually
+   being played. The bits parameter must be 8 or 16, freq is the sample
+   rate of the data, in Hertz. The vol and pan values use the same 0-255
+   ranges as the regular sample playing functions. The stereo parameter
+   should be set to 1 stereo streams, or 0 otherwise. If you want to
+   adjust the pitch, volume, or panning of a stream once it is playing,
+   you can use the regular voice_*() functions with stream-&gtvoice as a
+   parameter. The sample data is always in unsigned format, with stereo
+   waveforms consisting of alternating left/right samples, left sample
+   first.
 
 @@void @stop_audio_stream(AUDIOSTREAM *stream);
 @xref play_audio_stream