Mail Archives: djgpp/1997/03/24/23:46:27
From: | pv AT cs DOT montana DOT edu (Paul Peavyhouse)
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Newsgroups: | comp.os.msdos.djgpp
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Subject: | How to write a network "voice phone" program?
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Date: | Mon, 24 Mar 1997 18:49:22 GMT
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Organization: | Montana State University
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Lines: | 29
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Message-ID: | <5h7b0e$slj@netra.montana.edu>
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NNTP-Posting-Host: | esus.cs.montana.edu
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To: | djgpp AT delorie DOT com
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DJ-Gateway: | from newsgroup comp.os.msdos.djgpp
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This question is not necessarily a djgpp question, but I'd like to see
how djgpp can handle this idea. I'm thinking more along the lines of
Microsoft's Direct sound, but even as low-latency as they say their sound card
routines are, I have my doubts as to how well it can be implemented even in
DirectSound.
The idea is to write a barebones, no bells and whistles network "voice
phone" program. The idea is MINIMAL bandwidth and MINIMAL features. Voice
quality need not be anything better than you get on a typical CB or Walkie
Talkie.
So, my question is, does anyone know of any good libraries already
wriiten that allow line/mic input to a buffer (which I can then write the code
to transmit that buffer to another computer on a TCP/IP network) and then the
ability to echo that buffer to a lineout/speaker? This is a fairly
simple/common concept, so I thought I would check the global library before I
go about writing it myself. The trick though is that don't know much about
ANALOG input. Reading/writing digital WAVs and MODs is one thing, but to
process real-time analog input is over my head. Anyone have any suggestions
on the SIMPLEST way to do this? Should I do this in DJGPP (w/ ASM), or ignore
gnu and just use DirectSound?
PV
______________________________________________________________________________
Paul Peavyhouse
http://www.cs.montana.edu/~pv
email: pv AT cs DOT montana DOT edu
______________________________________________________________________________
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