Although the best recording conditions can't probably be achieved
recording directly to a computer (under Unix or some other operating
systems). We accept that in many cases recording directly to a
computer has many conveniences outweighing its disadvantages.
The disadvantages are primarily in quality, the electromagnetic noise
generated by a machine is large and most computers have particularly poor
shielding of there audio hardware such that noise always gets added to
the recorded signal. But there are ways to try to minimize this.
Getting better quality sound cards helps, but they can be very
expensive. "Professional" sound cards can go for as much as a
thousand dollars.
The advantage of using a computer directly is that you have much more
control over the recording session and first line processing can down at
record-time. In recent years we found the task of transferring the
recorded data from DAT tapes to a computer, and splitting them into
individual files, even before phonetic, labeled a significantly
laborious task, often larger and resource intensive than the rest
of the voice building process. So recently we've accept that direct
recording to disk using a machine worthwhile, except for voices that
require the highest quality (and when we have the money to take
more time). This section describes the issues in recording
under Unix, though they mostly apply under Windows too if you
go that route.
The first thing you'll find out about recording on a computer is that
no one knows how to do it, and probably no-one has actually used
the microphone on the machine at all before (or even knows if there
is a microphone). Although we believe we are living in a multi-media
computer age, setting up audio is still a tricky thing and even when
it works its often still flakey.
First you want to ensure that audio works on the machine at all. Find
out if anyone has actually heard any audio coming from it. Even though
there may be an audio board in there, it may not have any drivers
installed or the kernel doesn't know about it. In general, audio
rarely, "just works" under Linux in spite of people claiming Linux is
ready for the desktop. But before you start claiming Windows is better,
we've found that audio rarely "just works" there too. Under Windows
when it works its often fine, but when it doesn't the general Windows
user is much less likely to have any knowledge about how to fix it,
while at least in the Linux world, users have more experience in getting
recalcitrant devices to come to life.
Its difficult to name products here as the turn over in PC hardware is
frantic. Generally newer audio cards wont work and older card do. For
audio recording, we only require 16bit PCM, and none of the
fancy, FM synthesizers and wavetable devices, those are irrelevant and
often make card difficult or very hard to use. Laptops are particular
good for recording, as they generally add less noise to the signal
(especially if run on the battery) and they are portable enough to take
into a quite place that doesn't have desktop cooling fan running in the
background. However sound on Laptops under Unix (Linux, FreeBSD,
Solaris etc) is unfortunately even less likely to work, due to leading
edge technology and proprietary audio chips. Linux is improving in this
area but although we are becoming relatively good at getting audio to
work on new machines, its still quite a skill.
In general search the net for answers here. Linux offers both
ALSA and the Open Sound
drivers drivers which go a long way to help.
Note though even when these work there may be other problems (e.g. on
one laptop you can have either sound working or suspend working but not
both at once).
To test audio you'll need something to modify the basic gain on
the audio drivers (e.g. xmixer under Linux/FreeBSD, or
gaintool under Solaris). And you can test audio out
with the command (assuming you've set ESTDIR}
$ESTDIR/bin/na_play $ESTDIR/lib/example_data/kdt_001.wav
which should play a US male voice saying "She had your dark suit in
greasy washwater all year."
To test audio in you can use the command
$ESTDIR/bin/na_record -o file.wav -time 3
where the time given is the number of seconds to record. Note you may
need to change the microphone levels and/or input gain to make this work.
You should look at the audio signal as well as listen to it. Its
often quite easy to see noise in a signal than hear it. The human ear
has developed so that it can mask out noise, but unfortunately not
developed enough to mask all noise in synthesis. But in synthesis when
we are going to concatenated different parts of the signal the human ear
isn't as forgiving.
The following is a recording made with background noise, (probably a
computer).
The same piece of speech (re-iterated) in a quiet environment
looks like
As you can see the quiet parts of the speech are much quieter
in the clean case than the noise case.
Under Linux there is a Audio-Quality-HOWTO document that
helps get audio up and running. AT time of writting it
can be found http://audio.netpedia.net/aqht.html
