alsa::viavoice --- Moving to a minimalistic server

["T. V. Raman" <raman@xxxxxxxxxxx>]

When I wrote the ALSA-based TTS server, the sequence from memory
was:

0)      I implemented the server based on the aplay.c sources.
1)      After getting it working, I set up an asoundrc to use
DMix


It turns out that once you have a working asoundrc, some of the C
code in the server implementation that configures  the device is
unnecessary; I've now moved the redundant code into a #ifdef
DEBUG section with a view to eventually nuking it.

You can still compile in the old code by saying 
make -f makefile.debug atcleci.so 

Note that I've also moved  the code that sets up xfer align into
the debug section. That piece of code is 
relevant for cases where you see underflow/overflow problems ---
and the key params are start_delay and stop_delay --- if you see
underruns  --- this means that the sound buffer did not have
enough data in it from the TTS engine ---
I suggest playing with the start_delay value --- this might be
settable from the asoundrc as well.

-- 
Best Regards,
--raman

      
Email:  raman@xxxxxxxxxxx
WWW:    http://emacspeak.sf.net/raman/
AIM:    emacspeak       GTalk: tv.raman.tv@xxxxxxxxxxx
PGP:    http://emacspeak.sf.net/raman/raman-almaden.asc
Google: tv+raman 
IRC:    irc://irc.freenode.net/#emacs

-----------------------------------------------------------------------------
To unsubscribe from the emacspeak list or change your address on the
emacspeak list send mail to "emacspeak-request@xxxxxxxxxxx" with a
subject of "unsubscribe" or "help"