comp.speech Frequently Asked Questions - part 2/3
Found at http://www.jmas.co.jp/FAQs/comp-speech-faq
From: andrew.hunt@east.sun.com (Andrew Hunt)
Newsgroups: comp.speech,comp.answers,news.answers
Subject: comp.speech Frequently Asked Questions - part 2/3
Date: 12 Jul 1998 12:00:23 GMT
Approved: news-answers-request@MIT.Edu
Message-ID: <comp-speech-faq/part2_900244804@rtfm.mit.edu>
Archive-name: comp-speech-faq/part2
Last-modified: 1998/07/06
URL: http://www.speech.su.oz.au/comp.speech/
COMP.SPEECH FAQ POSTING - PART 2/3
[Note: this document has been automatically extracted from a WWW site:
http://www.speech.su.oz.au/comp.speech/
This may introduce some formatting errors.]
Signal Processing for Speech
comp.speech FAQ Section 2
* SpeechLinks: Signal Processing for Speech
* Q2.1: What sampling do I need for speech?
* Q2.2: Finding the pitch of a speech signal
* Q2.3: How do I find the start and end points of a speech
signal?
* Q2.4: Where can I find FFT software?
* Q2.5: Signal processing in speech technology
* Q2.6: Speech sampling and signal processing hardware
* Q2.7: How do I convert to/from mu-law format?
* Q2.8: Signal Processing Software
___________________________________________________________________________
Q2.1: What sampling do I need for speech?
For recorded speech to be understood by humans you need an 8kHz
sampling rate or more and at least 8 bit sampling. This produces poor
quality speech - but in can be understood.
Improvements can be achieved by increasing the number of bits in
sampling to 12bits or 16bits, or by using a non-linear encoding
technique such as mu-law or A-law (see Q2.7). This improves the
"signal-to-noise" ratio.
Increasing the sampling rate above 8kHz, say to 10kHz, 16kHz or 20Khz,
improves the frequency response: the higher the sampling frequency the
better the high frequency content will be. A 16kHz sampling rate is a
reasonable target for high quality speech recording and playback.
When doing speech recognition you need to remember that the your
computer is not as good as your ear so it will have trouble with poor
quality sounds. The choice of an appropriate sampling setup depends
very much on the speech recognition task and the amount of computer
power available.
___________________________________________________________________________
Q2.2: Finding the pitch of a speech signal
This topic comes up regularly in the comp.dsp newsgroup. Question 2.5
of the FAQ posting for comp.dsp gives a comprehensive list of
references on the definition, perception and processing of pitch. The
comp.dsp FAQ posting is posted regularly to the comp.dsp newsgroup,
and is also available by ftp and on the WWW:
* http://www.bdti.com/faq/dsp_faq.htm
* ftp://rtfm.mit.edu/pub/usenet/comp.dsp/
The following provide pitch tracking software:
* Most of the speech processing environments listed in Q1.9
including CSRE, ESPS, Kay Elemetrics Computer Speech Lab, OGI
Speech Tools, Speech Filing System, Signalyze, Soundscope.
___________________________________________________________________________
Q2.3: Finding start and end points of a speech signal
End-point detection algorithms identify sections in an incoming audio
signal that contain speech. Accurate end-pointing is a non-trivial
task, however, reasonable behaviour can be obtained for inputs which
contain only speech surrounded by silence (no other noises). Typical
algorithms look at the energy or amplitude of the incoming signal and
at the rate of "zero-crossings". A zero-crossing is where the audio
signal changes from positive to negative or visa versa. When the
energy and zero-crossings are at certain levels, it is reasonable to
guess that there is speech. More detailed descriptions are provided in
the papers cited below and in the documentation for the following
software.
End-point detection software is available from:
* ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/tools/ep.1.0.tar.gz
*
ftp://ftp.isip.msstate.edu/pub/software/signal_detector/sigd_v2.2.t
ar.gz
Plenty of research papers have been presented on end-pointing. Try the
following:
* Rabiner LR, Sambur MR, "An Algorithm for Determining the Endpoints
of Isolated Utterances", Bell System Technical Journal, Vol 54,
No. 2, pp 297-315, 1975.
* Drago, P.G. et al. "Digital Dynamic Speech Detectors." IEEE Trans
on Communications, Vol 26, No 1, Jan 78, pp. 140-145.
* Newman, W.C. "Detecting Speech with an Adapative Neural Network."
Electronic Design. 22 March 1990.
* Taboada. J et al "Explicit Estimation of Speech Boundaries" IEE
Proc. Sci. Meas. Technol., Vol 141, No.3, May 1994, pp 153-159.
___________________________________________________________________________
Q2.4: FFT Software
* Comprehensive list of FFT software
Links to over 65 different pieces of one-dimensional FFT code.
http://tjev.tel.etf.hr/josip/DSP/fft.html
* FFT Software including optimised fft routines and mixed-radix
algorithms
ftp://usc.edu/pub/C-numanal/fft-stuff.tar.gz
OR,
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/analysis/fft-stuff.
tar.gz
* mixfft03.zip:C-source for a very fast arbitrary N FFT routine
The C-source is ShareWare: read the text file included in the
package before using the FFT routine commercially.
Jens J. Nielsen: jnielsen@internet.dk
Available from
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/analysis/mixfft03.z
ip
OR ftp://ftp.coast.net/simtel/msdos/c/mixfft03.zip
* FFTW
FFTW is a C subroutine library for computing the FFT in one or
more dimensions. It is not limited to sizes that are powers of
two, and includes real-complex and parallel transforms.
Also on the FFTW web site are benchmarks comparing the
performance and accuracy of many public-domain FFT
implementations on a variety of platforms, as well as links to
other sources of FFT code and information.
Available from http://theory.lcs.mit.edu/~fftw
Developed by Matteo Frigo and Steven G. Johnson:
fftw@theory.lcs.mit.edu
___________________________________________________________________________
Q2.5: Signal processing in speech technology
This question is far to big to be answered in a FAQ posting. Here are
some WWW resources and books which cover the area well.
Tony Robinson's Course Notes
Dr. Tony Robinson of the Engineering Dept of Cambridge University has
put his Speech Analysis course notes on the web. The base page is
http://svr-www.eng.cam.ac.uk/~ajr/SA95/. There is information on the
following:
* Sampling theory
* Filter bank analysis
* Short-term fourier analysis
* Linear prediction analysis
* Formant analysis and voicing analysis
* Speech coding
* and more....
Joseph Picone's Course Notes
Joseph Picone of the Institute for Signal and Information Processing
(ISIP) at Mississippi State University has put two sets of course
notes on the web:
EE 4773/6773: Digital Signal Processing
The course covers sampling, frequency analysis, z-transforms,
filter design and more. The WWW site provides the syllabus,
assignments, some source code data, exams, homework and
solutions, lecture notes and more.
EE 8993: Fundamentals of Speech Recognition
The course covers background probability and
phonetics/acoustics, speech signal analysis, dynamic
programming, dynamic time warping, hidden Markov modelling,
language modelling, neural networks, etc. The WWW sites
provides the syllabus and lecture notes.
Signal Processing Home page
The Signal Processing Home page has information on a range of DSP
issues. It includes references to a range of software and much more.
http://tjev.tel.etf.hr/josip/DSP/sigproc.html
Books and other References
There are many good books which discuss signal processing for speech:
* Digital processing of speech signals; L. R. Rabiner, R. W.
Schafer. Englewood Cliffs; London: Prentice-Hall, 1978
* Voice and Speech Processing; T. W. Parsons. New York; McGraw Hill
1986
* Computer Speech Processing; ed Frank Fallside, William A. Woods
Englewood Cliffs: Prentice-Hall, c1985
* Digital speech processing : speech coding, synthesis, and
recognition edited by A. Nejat Ince; Kluwer Academic Publishers,
Boston, c1992
* Speech science and technology; edited by Shuzo Saito pub. Ohmsha,
Tokyo, c1992
* Speech analysis; edited by Ronald W. Schafer, John D. Markel, New
York, IEEE Press, c1979
* Applied Speech Technology Edited by: Ann Syrdal (AT&T Bell Labs,
Holmdel, New Jersey), Raymond Bennett (Ameritech, Hoffman Estates,
Illinois) and Steven Greenspan (AT&T Bell Labs, Murray Hill, New
Jersey). Publisher: CRC Press.
* Speech Communication: Human and Machine Douglas O'Shaughnessy,
Addison Wesley series in Electrical Engineering: Digital Signal
Processing, 1987.
* Discrete-time processing of speech signals; John R Deller, John G
Proakis, John H L Hansen; Macmillan 1993.
* Signal processing of speech; F J Owens; Macmillan 1993.
___________________________________________________________________________
Q2.6: Speech sampling and signal processing hardware
In addition to the following information, have a look at the Audio
File format document prepared by Guido van Rossum (see details in
Section 1.8).
Information is included on hardware for the following systems:
* Macintosh Audio Hardware
* PC Audio Hardware
* Unix Audio Hardware
Can anyone provide information for SGI, NeXT, other UNIX hardware and
any other PC soundcards?
Macintosh Audio Hardware - an overview
* Description: ALL Macintosh computers come with the ability to play
back sounds at any sample rate (sample rate conversion is done in
software.) Older machines have 8 bit stereo output (hardware runs
at 22254 samples/second). The newer machines have 16 bit stereo
hardare running at 44100 samples/second.
Most of the recent Macintosh computers come with sound input
hardware. There are probably exceptions to this, but the older and
some of the current low-end machines have 8 bit (linear) mono
hardware running at 22254.54 samples/second. All of the PowerPC,
AV, and the 500 series notebook computers come with 16 bit 44kHz
stereo sampling hardware. They can also record at 22050
samples/second. The sound manager implements an AGC (Automatic
Gain Control) function for the 8 bit hardware. The drivers have a
switch to turn off the AGC.
There are a number of DSP vendors that support high quality audio.
Generally this means quieter analog sections, and more IO formats
(AES/IBU, for example). Try DigiDesign and Spectral Innovations.
The software drivers for sound are described in "Inside Macintosh:
Sound". If you want to see some sample code check out the sources
for the Matlab "Sound and Image Toolbox". They can be found at
ftp://ftp.apple.com/pub/malcolm/SoundAndImageToolbox.cpt.
hqx
Routines that play and record sounds using the toolbox are
included (and interfaced to Matlab).
PC Audio Hardware
Note: new soundcards are becoming available all the time - the
information below is definitely not up to date. Check out the
following newsgroups for up-to-date information.
* comp.sys.ibm.pc.soundcard
* comp.sys.ibm.pc.soundcard.GUS
* comp.sys.ibm.pc.soundcard.advocacy
* comp.sys.ibm.pc.soundcard.games
* comp.sys.ibm.pc.soundcard.misc
* comp.sys.ibm.pc.soundcard.music
* comp.sys.ibm.pc.soundcard.tech
The Soundcard WWW Site is an excellent source of information:
* http://www.wi.leidenuniv.nl/audio/
An good source of programs and information for soundcards is SimTel:
* http://www.acs.oakland.edu/oak/SimTel/win3/sound.html
Additional information on PC soundcards is provided by the FAQ
postings for the comp.sys.ibm.pc.soundcard.misc newsgroup. These are
available by anonymous ftp from:
ftp://rtfm.mit.edu/pub/usenet/comp.sys.ibm.pc.soundcard.misc/
* Aria Soundcard FAQ
* Aria Soundcard Support List
* MIDI files software archives on the Internet
* Turtle Beach sound cards FAQ
Unix Audio Hardware
Could someone please provide information on the audio capabilities of
other Unix platforms?
Sun standard audio port: SPARC I & II
* Input and Output: 1 channel, 8 bit mu-law encoded, 8kHz sample
rate. This provides telephone quality sampling.
Sun DBRI audio port (SPARC 10 & 20)
* Input and Output: Stereo (2 channels). 16-bit linear sampling.
Multiple sample rates (48000, 44100, 37800, 32000, 22050, 18900,
16000, 11025, 9600, 8000 Hz)
Silicon Graphics Audio
The Silicon Graphics audio Frequently Asked Questions (FAQ) is the
best place to get information on SGI audio capabilities and
programming. It provides information on connecting the audio output,
using the DSP capabilities, controlling the audio output, programming,
useful software and more. It is available from:
* WWW: http://www-viz.tamu.edu/~sgi-faq/faq/html/audio/
* News: comp.sys.sgi.misc
* Ftp: ftp://viz.tamu.edu/pub/sgi/faq/
Ariel Signal Processors
* Platform: Various
* Description: A range of signal I/O, A/D, D/A and DSP products are
available. There are too many to list.
* Contact: Ariel Corp.
433 River Road, Highland Park, NJ 08904.
Ph: 908-249-2900 Fax: 908-249-2123 DSP BBS: 908-249-2124
___________________________________________________________________________
Q2.7: How do I convert to/from mu-law format?
Mu-law coding is a form of compression for audio signals including
speech. It is widely used in the telecommunications field because it
improves the signal-to-noise ratio without increasing the amount of
data. Typically, mu-law compressed speech is carried in 8-bit samples.
It is a companding technqiue. That means that carries more information
about the smaller signals than about larger signals.
On SUN Sparc systems have a look in the directory /usr/demo/SOUND.
Included are table lookup macros for ulaw conversions. [Note however
that not all systems will have /usr/demo/SOUND installed as it is
optional - see your system admin if it is missing.]
OR, here is some sample conversion code in C.
/**
** Signal conversion routines for use with Sun4/60 audio chip
**/
#include stdio.h
unsigned char linear2ulaw(/* int */);
int ulaw2linear(/* unsigned char */);
/*
** This routine converts from linear to ulaw
**
** Craig Reese: IDA/Supercomputing Research Center
** Joe Campbell: Department of Defense
** 29 September 1989
**
** References:
** 1) CCITT Recommendation G.711 (very difficult to follow)
** 2) "A New Digital Technique for Implementation of Any
** Continuous PCM Companding Law," Villeret, Michel,
** et al. 1973 IEEE Int. Conf. on Communications, Vol 1,
** 1973, pg. 11.12-11.17
** 3) MIL-STD-188-113,"Interoperability and Performance Standards
** for Analog-to_Digital Conversion Techniques,"
** 17 February 1987
**
** Input: Signed 16 bit linear sample
** Output: 8 bit ulaw sample
*/
#define ZEROTRAP /* turn on the trap as per the MIL-STD */
#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
#define CLIP 32635
unsigned char
linear2ulaw(sample)
int sample; {
static int exp_lut[256] = {0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,
4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7};
int sign, exponent, mantissa;
unsigned char ulawbyte;
/* Get the sample into sign-magnitude. */
sign = (sample >> 8) & 0x80; /* set aside the sign */
if (sign != 0) sample = -sample; /* get magnitude */
if (sample > CLIP) sample = CLIP; /* clip the magnitude */
/* Convert from 16 bit linear to ulaw. */
sample = sample + BIAS;
exponent = exp_lut[(sample >> 7) & 0xFF];
mantissa = (sample >> (exponent + 3)) & 0x0F;
ulawbyte = ~(sign | (exponent << 4) | mantissa);
#ifdef ZEROTRAP
if (ulawbyte == 0) ulawbyte = 0x02; /* optional CCITT trap */
#endif
return(ulawbyte);
}
/*
** This routine converts from ulaw to 16 bit linear.
**
** Craig Reese: IDA/Supercomputing Research Center
** 29 September 1989
**
** References:
** 1) CCITT Recommendation G.711 (very difficult to follow)
** 2) MIL-STD-188-113,"Interoperability and Performance Standards
** for Analog-to_Digital Conversion Techniques,"
** 17 February 1987
**
** Input: 8 bit ulaw sample
** Output: signed 16 bit linear sample
*/
int
ulaw2linear(ulawbyte)
unsigned char ulawbyte;
{
static int exp_lut[8] = {0,132,396,924,1980,4092,8316,16764};
int sign, exponent, mantissa, sample;
ulawbyte = ~ulawbyte;
sign = (ulawbyte & 0x80);
exponent = (ulawbyte >> 4) & 0x07;
mantissa = ulawbyte & 0x0F;
sample = exp_lut[exponent] + (mantissa << (exponent + 3));
if (sign != 0) sample = -sample;
return(sample);
}
___________________________________________________________________________
Q2.8: Signal Processing Software
[Note: Question 1.9 lists speech laboratory environments and audio
editors, many of which provide basic and advanced signal processing
capabilities.]
Signal Processing Products
* SigLib from Numerix Ltd.
On the Web
The following sites provide lists of useful DSP software. Not all the
software is directly applicable to speech processing.
comp.dsp FAQ
http://www.bdti.com/faq/dsp_faq.htm
DSP Internet Resources
http://www.eg3.com/
http://www.eg3.com/dsp.htm
Poynton's Digital Signal Processing Resource List
http://www.inforamp.net/~poynton/Poynton-dsp.html
WWW Pages Relating to Sound Computation
http://datura.cerl.uiuc.edu/netstuff/sigsoundLinks.html
Yahoo - Signal and Image Processing
http://www.yahoo.com/Science/Engineering/Electrical_Engineering
/Signal_and_Image_Processing/
Sound Related Resources
http://pscinfo.psc.edu/~geigel/menus/sound.html
SPLIB: Signal Processing url LIBrary
http://jazz.rice.edu/splib/
Wavelet's Home Page
http://www.mat.sbg.ac.at/~uhl/wav.html
SigLib from Numerix Ltd.
* Platform: Windows, Unix and all major DSPs
* Description: SigLib is an ANSI C Source DSP Library and includes
functions for the following areas : spectrum analysis, windowing,
filtering (fixed and adaptive coefficient), convolution,
correlation, covariance, signal generation, statistical analysis,
regression analysis, communications and modulation, digital
effects, vectors processing, control, graphics and file I/O.
Detailed product information and a description of the application
of SigLib to speech processing is provided on the Numerix Ltd. WWW
site.
* Availability: A free demonstration of SigLib V2.0 is available
from the Numerix Ltd. WWW site. Educational discount is available
for SigLib.
* Contact: Numerix Ltd.,
157 Sileby Road, Barrow-on-Soar, Leics, LE12 8LW, UK.
Phone/Fax : +44 (0)1509 413195
Email: numerix@numerix.co.uk
WWW: http://www.numerix.co.uk/
___________________________________________________________________________
Speech Coding and Compression
comp.speech FAQ Section 3
* SpeechLinks: Speech Coding
* Q3.1: Speech compression techniques
* Q3.2: Information on speech coding and compression
* Q3.3: Speech Compression / Coding Software
___________________________________________________________________________
Q3.1: Speech compression techniques
Provided by Tony Robinson:
The aim of speech compression is to produce a compact representation
of speech sounds such that when reconstructed it is perceived to be
close to the original. The two main measures of closeness are
intelligibility and naturalness.
The standard reference point is toll quality speech, this is the same
as what would be expected over a telephone line, for example, speech
coded at 8 kHz using 8 bit ulaw coding and a maximum frequency of
about 3.3 kHz. This is a bit rate of 64 kbps, and as such represents a
compressed form over (say) 16 bit, 16 kHz speech which is the standard
in speech recognition work.
ulaw coding does not exploit the (normally large) sample to sample
correlations found in speech. ADPCM is the next family of speech
coding techniques, and does exploit this redundancy by using a simple
linear filter to predict the next sample of speech. The resulting
prediction error is typically quantised to 4 bits thus giving a bit
rate of 32 kbps (see, for example, the software in Q3.3: 32 kbps
ADPCM, G.711/721/723 Compression, shorten). The advantages of ADPCM
are that is simple to implement and has very low delay.
To obtain more compression specific properties of the speech signal
must be modelling. The main assumption is known as the source filter
model of speech production. This assumes that a source (voicing or
fricative excitation) is passed through a filter (the vocal tract
response) to produce the speech. The simplest implementation of this
is known as a LPC synthesiser (e.g. LPC10e). At every frame the speech
is analysed to compute the filter coefficients, the energy of the
excitation, a voicing decision, and a pitch value if voiced. At the
decoder a regular set of pulses for voiced speech or white noise for
unvoiced speech is passed through the linear filter and multiplied by
the gain to produce the speech. This is a very efficient system and
typically produces speech coded at 1200-2400bps. With clever acoustic
vector prediction this can be reduced to 300-600bps. The disadvantages
are a loss of naturalness over most of the speech and occasionally a
loss of intelligibility.
The CELP family of coders compensates for the lack of quality of the
simple LPC model by using more information in the excitation. Each of
a set of codebook of excitation vectors is tried and the index of the
one that best matches the original speech is transmitted. This results
in an increase in the bit rate to typically 4800-9600bps. Most speech
coding research is currently directed towards CELP coders. (See, for
example, CELP 3.2a, a TMS implementation, a G.728 LD-CELP vocoder, and
the L&H implementation.
___________________________________________________________________________
Q3.2: Information on speech coding and compression
Reference Books
The following books cover speech coding/compression.
* Douglas O'Shaughnessy, Speech Communication: Human and Machine,
Addison Wesley series in Electrical Engineering: Digital Signal
Processing, 1987.
* Bishnu Atal in ed. Fallside, F. and W. Woods, ed. Computer Speech
Processing. London: Prentice/Hall International, 1985. N. S.
Jayant and P. Noll, Digital Coding of Waveforms, Prentice Hall,
ISBN 0-13-211913-7 01, 1984.
* W.B. Kleijn and K.K. Paliwal (Eds.), Speech Coding and Synthesis,
Elsevier, Amsterdam, 1995.
Contents, preface etc on the WWW:
http://www.elsevier.nl/section/engtech/scs/menu.htm
* Thomas P. Barnwell, Kambiz Nayebi and Craig H Richardson, Speech
Coding: A Computer Laboratory Textbook, John Wiley and Sons Inc,
1996.
* Schuyler R Quackenbush, Tom P Barnwell III, Mark A Clements,
Objective Measures of Speech Quality, Prentice-Hall, 1988.
And the are good tutorial articles.
* Makhoul, J. "Linear Prediction: A Tutorial Review." Proc. of the
IEEE 63 (1975): 561 - 580.
On the WWW
comp.compression FAQ
Includes a few questions and answers on the compression of
speech.
ftp://rtfm.mit.edu/pub/usenet/comp.compression/
Tony Robinson's Speech Analysis Course
A complete course on speech analysis, including some stuff on
speech coding.
http://svr-www.eng.cam.ac.uk/~ajr/SA95/
http://svr-www.eng.cam.ac.uk/~ajr/SA95/node78.html
ITU Coding Standards
Members of the ITU (International Telecommunications Union) can
obtain copies of the Series G Recommendations (including
G.711/721/723/728) from the ITU WWW site (http://www.itu.ch/)
and from http://www.itu.ch/itudoc/itu-t/rec/g/g700-799.html.
Jason Woodard's Speech Coding Page
Introduction to speech coding plus information on a series of
speech coding standards.
http://www-mobile.ecs.soton.ac.uk/speech_codecs/index.html
WWW searchable online-bibiliography for Phonetics and Speech
Technology
Over 8000 entries provided by Institut fur Phonetik at Johann
Wolfgang Goethe-Universitat Frankfurt.
http://www.uni-frankfurt.de/~ifb/bib_engl.html
Ciaran McElroy's Speech Coding Page
Introduction to many types of speech coding.
http://wwwdsp.ucd.ie/speech/tutorial/speech_coding/speech_tut.h
tml
Examples of speech coding
Nam Phamdo's Speech Coding Demonstration
Examples of ADPCM, LD-CELP, CELP, LPC10 and CELP coding and
coding over a noisy channel.
http://admii.arl.mil/~fsbrn/phamdo/speech_demo.html
Phil Karn's Digital/Analog Voice Demo
Examples of several speech coding systems.
http://www.qualcomm.com/people/pkarn/voicedemo/
___________________________________________________________________________
Q3.3: Speech Compression / Coding Software
The following speech compression software is described in the FAQ.
* 32 kbps ADPCM
* Castleton Network Systems - G.729 Voice Coder
* CELP 3.2a & LPC-10
* 8 Kbit/s CELP on the TMS320C5x family of DSP chips
* CyberVoice
* Rockwell's DigiTalk
* File format conversion
* G.711/721/723 Compression
* G.728 LD-CELP vocoder
* G.728 Compression
* GSM 06.10 Compression
* Lernout & Hauspie Speech Coding (5 products)
* Lernout & Hauspie Speech Coding SDK
* MPEG Audio
* shorten - a lossless compressor for speech signals
* Sipro Lab Telecom Inc. Coding
* Sonarc: Digital Audio Compression
* StarAudio Compressor/Player
* TrueSpeech from DSP Group
* U.S.F.S. 1016 CELP vocoder for DSP56001
* ToolVox from Voxware
32 kbps ADPCM
* Platform: SGI and Sun Sparcs
* Description: 32 kbps ADPCM C-source code (G.721 compatibility is
uncertain)
* Contact: Jack Jansen
* Availablity: http://www.cwi.nl/ftp/audio/adpcm.shar
Castleton Network Systems - G.729 Voice Coder
* Platform: TI TMS320C5x DSP
* Description: G.729, also called CS-ACELP (Conjugate-Structure
Algebraic Code Excited Linear Prediction), is a state-of-the-art
voice compression ITU (International Telecommunications Union)
standard that can be used in a wide range of applications
including wireless communications, digital satellite systems,
packetized speech and digital leased lines. G.729 provides 8000
bits/s bandwidth for compressed speech at toll quality (equivalent
to G.726 32 kbit/s ADPCM under clean channel condition). Also,
G.729 has lower complexity and lower bit rate than G.728.
The Castleton G.729 implementation provides a bit-exact
implementation of the ITU standard on a single TI TMS320C5x DSP.
The software is C callable and fully re-entrant, which allows easy
interfacing and multi-channel capability. The encoder and decoder
are fully independent, therefore, a DSP device can run a number of
full-duplex or half-duplex channels. The coder and the decoder are
able to operate under a real-time task switching kernel.
* Cost and Availablity: Contact Castleton Network Systems.
* Contact: Castleton Network Systems Corporation
350 Terry Fox Drive, Kanata, Ontario, Canada K2K 2W5
Ph: 613-591-8786, Fax: 613-591-8783
Email: inquire@castleton.com
WWW: http://www.castleton.com/
CELP 3.2a & LPC-10
* Platform: Sun (the makefiles and source can be modified for other
platforms)
* Description: CELP is lossy compression technqiue. The US
Department of Defences's Federal-Standard-1016 based 4800 bps code
excited linear prediction voice coder version 3.2a (CELP 3.2a).
Fortran and C simulation source codes.
* Availability: By anonymous ftp from:
ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z
Or from the comp.speech ftp server
ftp://svr-ftp.eng.cam.ac.uk/comp.speech/coding/celp_3.2a.tar.Z
ftp://svr-ftp.eng.cam.ac.uk/comp.speech/coding/celp_3.2a.tar.gz
LPC-10 Fortran source code is also available:
ftp://ftp.super.org/pub/speech/lpc10-1.0.tar.gz
Here is a modified LPC-10 release that includes ANSI C source:
http://www.arl.wustl.edu/~jaf/lpc/
* Documentation: The following articles describe the
Federal-Standard-1016 4.8-kbps CELP coder:
+ Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C.
Welch, "The Federal Standard 1016 4800 bps CELP Voice Coder,"
Digital Signal Processing, Academic Press, 1991, Vol. 1, No.
3, p. 145-155.
+ Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C.
Welch, "The DoD 4.8 kbps Standard (Proposed Federal Standard
1016)," in Advances in Speech Coding, ed. Atal, Cuperman and
Gersho, Kluwer Academic Publishers, 1991, Chapter 12, p.
121-133.
The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400
bps linear prediction coder (LPC-10) was republished as a Federal
Information Processing Standards Publication 137 (FIPS Pub 137).
It is described in:
+ Thomas E. Tremain, "The Government Standard Linear Predictive
Coding Algorithm: LPC-10," Speech Technology Magazine, April
1982, p. 40-49.
There is also a section about FS-1015 in the book:
+ Panos E. Papamichalis, Practical Approaches to Speech Coding,
Prentice-Hall, 1987.
The voicing classifier used in the enhanced LPC-10 (LPC-10e) is
described in:
+ Campbell, Joseph P., Jr. and T. E. Tremain, "Voiced/Unvoiced
Classification of Speech with Applications to the U.S.
Government LPC-10E Algorithm," Proceedings of the IEEE Intl.
Conf. on Acoustics, Speech, and Signal Processing, 1986, p.
473-6.
* Vendors:
Realtime DSP code for FS-1015 and FS-1016 is sold by:
+ John DellaMorte, DSP Software Engineering
165 Middlesex Tpk, Suite 206, Bedford, MA 01730, USA
Ph: 1-617-275-3733 Fax: 1-617-275-4323
Email: dspse.bedford@channel1.com
DSP Software Engineering's FS-1016 code can run on a DSP
Research's Tiger 30 (a PC board with a TMS320C3x and analog
interface suited to development work).
+ DSP Research
1095 E. Duane Ave, Sunnyvale, CA 94086, USA
Ph: (408)773-1042 Fax: (408)736-3451
8 Kbit/s CELP on the TMS320C5x family of DSP chips
* Description: For low bandwidth transmission of voice, compact
voice storage for archival purposes, low-cost digital answering
machines and efficient storage for voice mail. Features :
+ near toll quality at 8 Kb/s.
+ Variable rate option with 1 Kb/s silence encoding.
+ Implemented on a fixed-point processor for lower system cost.
+ Attractive licensing scheme.
+ Future availability of 4 Kb/s.
+ Custom rates possible.
Capacity :
+ Two half-duplex or one full duplex channels on the 20 MIPS
'C5x (at 95% and 55% CPU utilization respectively).
+ Two full duplex channels on the 28.6 MIPS 'C5x (at 77% CPU
utilization).
+ Requires 9 K-words program memory and 3 K-words data memory.
+ Decoding in real-time on a 486 class CPU.
* Contact:
CVI Inc.
443 Vienna Cres. North Vancouver, BC, Canada V7N 3B3
Tel: (604) 987 1719 Fax: (604) 986 8139
Email: cvi@extropia.wimsey.com
CyberVoice
* Description: Cybernetics InfoTech, Inc. offers the following
products
+ Telephone voice compression at 1.2, 2.4, 4.8 and 6.0 kbit/s
with good-communications-quality to near-toll-quality coded
voice;
+ Wideband voice (7-kHz bandwidth) compression at 16 kbit/s
with near-original-quality coded voice;
+ Internet Voice E-mail software with voice editing,
high-quality low-data-rate voice compression, fast/slow voice
playback, and more.
* Availablity: C code and Windows .DLL for telephone voice
compression and wideband voice compression are available for
licensing.
Real-time DSP codes are under development.
Voice E-mail software is available for purchase and download from
the CyberVoice home page.
* Contact: Cybernetics InfoTech, Inc.
2 Professional Dr., #228, Gaithersburg, MD 20879
WWW: http://www.cybit.com/
E-mail: info@cybit.com
Fax: 301-590-0359
Rockwell's DigiTalk
* Description: The DigiTalk coder operates at a sampling rate of
8KHz and transmits 223 bits of coded speech every 26ms, giving an
overall bit rate of 8.577Kbps. The algorithm is based on
analysis-by-synthesis predictive coding with vector-coded
excitation, in which the excitation signal is optimized by
minimizing the perceptually weighted error between the original
and synthesized speech. More information and results of perceptual
tests are available on the WWW.
* Availablity: See the WWW page:
http://www.nb.rockwell.com/ref/digitalk/
File format conversion
* Platform: SUN OS?
* Description: Conversion utility able to encode and decode between
the the following formats: G.723, G.721, A-law, u-law and linear.
* Availability: By anonymous ftp from
ftp://ftp.cwi.nl/pub/audio/ccitt-adpcm.tar.Z
G.711/721/723 Compression
* Description:
+ G.711 : CCITT u-law and A-law compression
+ G.721 : CCITT 32 kbps ADPCM coder
+ G.723 : CCITT 24 kbps and 40 kbps ADPCM coders
* Availability: By email to itudoc@itu.ch, with
GET ITU-3022
as the *only* line in the body of the message.
It is also available by anonymous ftp from:
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/G711_G
721_G723.tar.Z
G.728 LD-CELP vocoder
* Platform: Analog Devices ADSP-2171
* Description: Real-time, full-duplex G.728 LD-CELP vocoder that
runs on a single Analog Devices ADSP-2171. Source and object code
available for a one-time license fee.
* Contact:
Cole Erskine
Analogical Systems
299 California Avenue, Suite 120
Palo Alto, CA 94306, USA
Tel:(415) 323-3232 FAX:(415) 323-4222
email: cole@analogical.com
G.728 Compression
* Description: G.728 low delay celp package written by Alex Zatsman
of Analog Devices, Inc.
* Availability: By anonymous ftp from
ftp://dspsun.eas.asu.edu/pub/speech/ldcelp.tgz
GSM 06.10 Compression
* Platform: Unix; faster than real time on most Sun SPARCstations
* Description: GSM 06.10 is a standardized lossy speech compression
employed by most European wireless telephones. It uses RPE/LTP
(residual pulse excitation/long term prediction) coding to
compress frames of 160 13-bit samples (8 kHz sampling rate, i.e. a
frame rate of 50 Hz) into 260 bits.
* Contact: GSM 06.10 support and implementation
_jutta@cs.tu-berlin.de_, cabo@cs.tu-berlin.de
* Availability: The following configurations are available be
anonymous ftp:
gzip compression from Germany:
ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/gsm-1.
0.7.tar.gz
MS-DOS compression from Germany:
ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/ddj/gs
m-107.zip
MS-DOS compression from USA:
ftp://ftp.mv.com/pub/ddj/1194.12/gsm-105.zip
* Misc: The WWW site is
http://www.cs.tu-berlin.de/~jutta/toast.html
Lernout & Hauspie Speech and Music Coding Product Range
* Product name: L&H.smc650: 32kbps ADPCM Speech coding
+ Implementation of ADPCM 32 kbps based on CCITT G721 standard.
+ Estimated quality: 4.1 MOS (Mean Opinion Score)
+ Hardware Example: Analog Devices ADSP2101
+ Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear
signal with up to 16 bits per sample; 8 kHz sampling rate
* Product name: L&H.smc550: LD-CELP 16 kbps speech coding
+ Proprietary implementation of LD-CELP 16 kbps based on CCITT
G728 standard.
+ Estimated quality: 4.0 MOS (Mean Opinion Score)
+ Hardware Example: Motorola 5600X
+ Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear
signal with up to 16 bits per sample; 8 kHz sampling rate
* Product name: L&H.smc450: 16-17.5 kbps speech coding
+ Estimated Quality: 3.9 MOS (Mean Opinion Score)
+ Hardware Examples: Analog Devices ADSP2101, Intel 486 DX2/66
MHz
+ Input / Output Signal: A-Law or mu-Law PCM (64 kbps); Linear
signal with up to 16 bits per sample; 8 kHz sampling rate.
* Product name: L&H.smc350: 4.8-9.6 kbps speech coding
+ Proprietary CELP based software for compression rates of 4.8
kbps to 9.6 kbps
+ Estimated Quality: 3.5 MOS (Mean Opinion Score)
+ Hardware Examples: AT&T DSP32C
+ Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear
signal with up to 16 bits per sample; 8 kHz or 11.025kHz
sampling rate.
* Product name: L&H.smc250: 2.4 kbps speech coding
+ Combination of multi band excitation and code book excited
linear prediction.
+ Estimated Quality: 3.0 MOS (Mean Opinion Score).
+ Hardware Examples: Intel 486 DX2/66 MHz, Analog Devices
ADSP2101
+ Input signal: A-Law or mu-Law PCM (64 kbps); Linear signal
with 12-15 bits per sample; 8 kHz sampling rate.
+ Output signal: A-Law or mu-Law PCM (64 kbps); Linear signal
with 12-15 bits per sample; 8 kHz sampling rate.
* See also: L&H Speech Coding SDK
* More Information: On the WWW: http://www.lhs.com/coding.html
* Cost: Unknown
* Contact: Lernout and Hauspie Speech Products
20 Mall Road, 4th Floor
Burlington, MA 01803, USA
Ph: +1-617-238-0960, Fax: +1-617-238-0986
Email: sales@lhs.com
WWW: http://www.lhs.com/
Lernout & Hauspie Speech Coding SDK
* Description: Windows based software development kit for
integrating speech coding technology with Windows based PC
applications.
* Requirements: IBM-compatible 486 DX/33 MHz + 2MB RAM + MS DOS 5.0
+ MS Windows 3.1 (or higher) + Sound Blaster compatible sound
board.
* See also: L&H Speech Coding Products
* More Information: On the WWW: http://www.lhs.com/coding.html
* Cost: Unknown
* Contact: Lernout and Hauspie Speech Products
20 Mall Road, 4th Floor
Burlington, MA 01803, USA
Ph: +1-617-238-0960, Fax: +1-617-238-0986
Email: sales@lhs.com
WWW: http://www.lhs.com/
MPEG Audio
MPEG (Moving Pictures Experts Group) is a standard methods for
compression and transmission of digital video and audio. Detailed FAQs
and WWW sites are available for MPEG:
MPEG Pointers and Resources
http://www.mpeg.org/
FAQ by Luigi: http://www.crs4.it/~luigi/MPEG/mpegfaq.html
FAQ by Frank Gadegast
http://www.powerweb.de/mpeg/mpegfaq/
FAQ by by Chad Fogg
http://www-plateau.cs.berkeley.edu/mpegfaq/MPEG-2-FAQ.html
How to Install an MPEG Audio Player for your Web Navigator
http://www.mpeg.org/index.html/MPEG-audio-player.html
MPEG Audio Software on the WWW
Audio and Music Applications for Silicon Graphics Systems
Lists 4 MPEG audio players for SGI machines.
http://reality.sgi.com/employees/cook/audio.apps/public.html
MPEG-1 Audio Layer 3 encoder, decoder and FAQ
From the Fraunhofer Institute
http://www.iis.fhg.de/departs/amm/layer3/index.html
MPEG-2 Audio FAQ from Philips
http://www.keymodules.philips.com/MD/mpeg/faqmpeg2.htm
MPEG-1 and MPEG-2 audio software
Universitaet Hannover
ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/
MPEG-1 Audio Layer 1 &2 encoder - decoder
Internet Underground Music Archive (IUMA)
ftp://ftp.iuma.com/audio_utils/converters/source/
Buddy Software Library: MPEG-1 Audio Layer 3 encoder and
player
http://www.buddy.org/softlib.html
MPEG-1 Audio Layer 1 & 2 decoder and verifier at CCETT
ftp://ftp.ccett.fr/pub/mpeg/audio_new/
MPEG-2 Audio encoder and decoder at CCETT
ftp://ftp.ccett.fr/pub/mpeg/mpeg2/
MPEG Audio - MetaSound
* Platform: MS Windows/3.1 and Windows/95
* Description: MetaSound is a partial MPEG-1 software decoder which
is designed to work with hardware video decoders. It can reduce
the hardware cost by eliminating the need for a hardware audio
decoder. Currently, MetaSound has been successfully incorporated
to work with three hardware video decoders. Features
+ Performance: For 486 DX4-100 machines or above, MetaSound can
deliver FM quality (22 KHz) sound. For Pentium-90 or above
machines, MetaSound requires 40% CPU bandwidth to deliver CD
quality (44.1 KHz) sound.
+ Portability: it can take less than one month to port to new
hardware video decoders.
+ CD standard supports including Video CD 1.0, Video CD 2.0,
and CDI.
+ User interface with full set of functions: volume control,
stop, pause, forward, backward, mute, resume, select the
previous/next program track (Video CD 2.0), randomly select a
program track (Video CD 2.0).
+ Error Recovery: can automatically skip error bitstreams.
* Contact: Meta Media, Inc.
F8, #10-1, Ho-Ping East Rd. Sec. 1, Taipei, Taiwan, R.O.C.
Ph: 011-886-2-369-3330, Fax: 011-886-2-369-3331
Email: mmedia@ms4.hinet.net.tw
shorten - a lossless compressor for speech signals
* Platform: UNIX/DOS
* Description: A fast waveform coder suitable for a speech and music
signals in a wide variety of file formats. The degree of
compression is adjustable from lossless to three bits a sample.
16bit 16kHz speech generally attains 50% lossless compression and
16:3 compression of CDROM quality speech is obtainable with only
minor audiable degredation.
* Availability: Anonymous ftp - UNIX and DOS versions
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte
n.tar.gz
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte
n.tar.Z
ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte
n.zip
Sipro Lab Telecom Inc. Coding
* Platform: Various processors
* Description: Coding software for several International Standards
plus two Proprietary standards.
International Standards
1. PCS 1900 (a 13 kbps codec, established as a North American
PCS standard)
2. Enhanced GSM (a 13 kbps codec)
3. G.723 (8 kbps codec established as a multi-purpose
international standard)
4. G.729 (a dual-rate codec for the video phone market)
5. G.729 Annex A (8 kbps codec made for Digital Simultaneous
Voice & Data transmission in the modem industry).
Proprietary Standards
1. ACELP 8 v2.0 codec (flexible dual rate codec equipped with a
VAD)
2. ACELP 4.8 codec
* Contact: Sipro Lab Telecom Inc.
770, Chemin Lucerne, Ville Mont-Royal (Quebec), H3R 2H6 CANADA
Ph: (514) 737-5874, Fax: (514) 737-2327
E-mail: sales@sipro.com
WWW: http://www.sipro.com/
Sonarc: Digital Audio Compression
* Platform: DOS and Windows
* Description: Sonarc provides reversable, variable-rate compression
of audio signals. Obtains compression ratio which averages about
2:1. Supports monaural and stereo files, 8-bit and 16-bit files,
and WAVE and VOC formats.
* Availablity: Shareware by Richard P. Sprague
Speech Compression
P.O. Box 1785, Wilsonville, OR, 97070-1785, USA
Ph: (503) 263-3102
Email: 76635.3652@compuserve.com
StarAudio Compressor/Player
* Platform: Win95
* Description: Using a time-domain process delivers lossless
decompressed data. Processes any source of .wav file format, high
quality 16-bit audio data at any sampling rate. Requires no
special hardware and decompression speed is real-time on most
486's and on any Pentium. The higher the sampling rate the higher
the compression ratio; minimum compression of 4:1 for 11k data,
and usually exceeding 7:1 for 44k data. Full bandwidth of signal
is preserved with default compression options. Compression options
allow increase of compression ratio further with a slight trade
off in the reduction of the output quality. A decompression
library is available for application development.
* Demo: Download the shareware version of the program from the STR
WWW site.
* Misc: A technical paper is available in Word 6.0 format:
ftp://ftp.speechtech.com/pub/speechtech/docs/audocw60.exe
* Contact: Speech Technology Research Ltd.,
Suite B - 1623 McKenzie Avenue, Victoria, B.C. V8N 1A6, Canada
Ph: +1-250-477-0544
Email: products@speechtech.com
WWW: http://www.speechtech.com/home/speechtech/
TrueSpeech from DSP Group
* Description: TrueSpeech is a family of speech compression and
decompression algorithms and software. It is designed for personal
computers and personal communications devices. With the high
compression ratios ranging from 15:1 to 27:1, TrueSpeech improves
the storage and communications transmission of digital voice
information and can be used in the integration of personal
computers and telephones. TrueSpeech can be utilized in many
products and applications such as:
+ Multimedia PCs
+ Sound cards and modems
+ Computer/telephony and teleconferencing
+ Voice mail systems and PBX systems
+ Wireless/cellular applications
+ Personal digital assistants
+ Games, Education
+ Video/cable and on-line services
The TrueSpeech encoder is available for free in the Sound System
of Windows 95 and Windows NT. The DSPG WWW pages have information
on how to add TrueSpeech capability to your WWW pages.
* Contact: DSP Group, Inc.
3120 Scott Boulevard, Santa Clara, CA 95054-3317, USA
Phone: (408) 986-4300 Fax: (408) 986-4323
Email: Webster@dspg.com
WWW: http://www.dspg.com/index.html
U.S.F.S. 1016 CELP vocoder for DSP56001
* Platform: DSP56001
* Description: Real-time U.S.F.S. 1016 CELP vocoder that runs on a
single 27MHz Motorola DSP56001. Free demo software available for
PC-56 and PC-56D. Source and object code available for a one-time
license fee.
* Contact:
Cole Erskine
Analogical Systems
299 California Avenue, Suite 120
Palo Alto, CA 94306, USA
Tel:(415) 323-3232 FAX:(415) 323-4222
Email: cole@analogical.com
ToolVox from Voxware
* Platform: Windows and soon available on Mac (in Beta now) and Unix
* Description: ToolVox is a proprietary frequency domain speech
coder. 11 KHz speech is coded to an average rate of between 5,000
bits per second and 9,000 bps. Real-time compression algorithms
available for 2,400 bps. 22 KHz playback, as well as a ultra low
bit rate 8 KHz codec are coming soon. On playback, the time scale
can be changed by a 5x factor, pitch can be modified over a 3
octave range, and vocal personality can be modified using a
tranformation function called VoiceFonts(tm).
* Misc 1: A SDK for Windows is available.
* Misc 2: Demo software is available from the Voxware Inc WWW page:
http://www.voxware.com/
* Price: Basic toolkit is $895 US. OEM and mass distribution
licenses are separate. Ordering information is provided on the
Voxware WWW server.
* Contact:
Voxware, Inc.
Ph: (609) 497-1212 Fax: (609) 497-2490
Sale information: sales@voxware.com
WWW: http://www.voxware.com/
___________________________________________________________________________
Natural Language Processing
comp.speech FAQ Section 4
There is now a newsgroup specifically for Natural Language Processing;
comp.ai.nat-lang. A FAQ posting is available for the group:
ftp://rtfm.mit.edu/pub/usenet/comp.ai.nat-lang/Natural_Language
_Processing_FAQ
There is also a lot of useful information on Natural Language
Processing in the comp.ai FAQ. That FAQ lists available software and
useful references. It includes a substantial list of software,
documentation and other info available by ftp.
The FAQ has information on the following:
* Q4.1: NLP References and Books
* Q4.2: NLP Software
___________________________________________________________________________
Q4.1: NLP References and Books
Take a look at the FAQ for the "comp.ai" newsgroup as it also includes
some useful references.
* James Allen: Natural Language Understanding, (Benjamin/Cummings
Series in Computer Science) Menlo Park: Benjamin/Cummings
Publishing Company, 1987.
+ This book consists of four parts: syntactic processing,
semantic interpretation, context and world knowledge, and
response generation.
* G. Gazdar and C. Mellish, Natural Language Processing in Prolog,
Addison Wesley, 1989
* G. Gazdar and C. Mellish, Natural Language Processing in Lisp,
Addison Wesley, 1989
* G. Gazdar and C. Mellish, Natural Language Processing in Pop11,
Addison Wesley, 1989
+ Emphasis on parsing, especially unification-based parsing,
lots of details on the lexicon, feature propagation, etc.
Fair coverage of semantic interpretation, inference in
natural language processing, and pragmatics; much less
extensive than in Allen's book, but more formal. There are
three versions, one for each programming language listed
above, with complete code.
* Shapiro, Stuart C.: Encyclopedia of Artificial Intelligence Vol.1
and 2. New York: John Wiley & Sons, 1990.
+ There are articles on the different areas of natural language
processing which also give additional references.
* Paris, Ce'cile L.; Swartout, William R.; Mann, William C.: Natural
Language Generation in Artificial Intelligence and Computational
Linguistics. Boston: Kluwer Academic Publishers, 1991.
+ The book describes the most current research developments in
natural language generation and all aspects of the generation
process are discussed. The book is comprised of three
sections: one on text planning, one on lexical choice, and
one on grammar.
* Readings in Natural Language Processing, ed by B. Grosz, K. Sparck
Jones and B. Webber, Morgan Kaufmann, 1986
+ A collection of classic papers on Natural Language
Processing. Fairly complete at the time the book came out
(1986) but now seriously out of date. Still useful for ATN's,
etc.
* Klaus K. Obermeier, Natural Language Processing Technologies in
Artificial Intelligence: The Science and Industry Perspective,
Ellis Horwood Ltd, John Wiley & Sons, Chichester, England, 1989.
The following are extensive bibliographies related to NLP:
* Computational Parsing : Syntactic Analysis, Semantic Analysis,
Semantic Interpretation, Parsing Algorithms, Parsing Strategies :
BIBLIOGRAPHY, by Conrad F. Sabourin 1994, 2 volumes, 1029p, ISBN
2-921173-02-6, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal,
H3X 3T4, Canada.
* Computational Text Understanding : Natural Language Programming,
Argument Analysis : BIBLIOGRAPHY, by Conrad F. Sabourin 1994,
657p, ISBN 2-921173-06-9, INFOLINGUA inc., P.O. Box 187 Snowdon,
Montreal, H3X 3T4, Canada.
See also: http://gomer.mlink.net/infolingua.html
* Computational Text Generation : Generation from data or Linguistic
Structure, Text Planning, Sentence Generation, Explanation
Generation : BIBLIOGRAPHY, by Conrad F. Sabourin with a survey
article by Mark T. Maybury 1994, 649p, ISBN 2-921173-07-7,
INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada.
See also: http://gomer.mlink.net/infolingua.html
* Natural Language Processing : Interfaces to Databases, to Expert
Systems, to Robots, to Operating Systems, and to
Question-Answering Systems : BIBLIOGRAPHY, by Conrad F. Sabourin,
1994, 2 volumes, 847p, ISBN 2-921173-08-5 INFOLINGUA inc., P.O.
Box 187 Snowdon, Montreal, H3X 3T4, Canada
See also: http://gomer.mlink.net/infolingua.html
Journals
The major journals of the field are
* Computational Linguistics and _Cognitive Science_ for the
artificial intelligence aspects,
* Cognition for the psychological aspects,
* Language and _Linguistics and Philosophy_ and Linguistic Inquiry
for the linguistic aspects.
* Artificial Intelligence occasionally has papers on natural
language processing.
Conferences
The major NLP conferences are
* ACL: held annually
* COLING: held biannually
Most AI conferences have a NLP track; AAAI, ECAI, IJCAI and the
Cognitive Science Society conferences usually interesting for NLP.
CUNY is an important psycholinguistic conference. Other conferences
include NELS, the conference of the Chicago Linguistic Society (CLS),
WCCFL, LSA, the Amsterdam Colloquium, and SALT.
___________________________________________________________________________
Q4.2: NLP Software
Natural Language Software Registry (NLSR) - NLP Tools
* The Natural Language Software Registry is available from the
German Research Institute for Artificial Intelligence (DFKI) in
Saarbrucken. Its purpose is to facilitate the exchange and
evaluation of natural language processing software within the
research community. To this end, the NLSR is cataloging natural
language software projects, both commercial and non- commercial.
The new updated and enlarged version contains more than 100
descriptions of natural processing software. Registry listings
include:
+ speech signal processors, such as the Computerized Speech Lab
(Kay Elemetrics)
+ morphological analyzers, such as PC-KIMMO (Summer Institute
for Linguistics)
+ parsers, such as Alveytools (University of Edinburgh)
+ semantic and pragmatic analyzer, such as NLL (University of
the Saarland, Germany)
+ generation programs, such as FUF (Ben Gurion University of
the Negev)
+ knowledge representation systems, such as Rhet (University of
Rochester)
+ multicomponent systems, such as ELU (ISSCO), PENMAN (ISI),
Pundit (UNISYS), SNePS (SUNY Buffalo),
+ NLP-Tools, such as GULP (University of Georgia) or Linguist
(Kansai Research Laboratory)
+ applications programs (misc.)
* If you have developed a piece of software for natural language
processing that other researchers might find useful, you can
include it by returning the questionnaire available from the
sources below.
* ftp://ftp.dfki.uni-sb.de/pub/registry
* e-mail: registry@dfki.uni-sb.de
* Natural Language Software Registry
Deutsches Forschungsinstitut fuer Kuenstliche Intelligenz (DFKI)
Stuhlsatzenhausweg 3
D-66123 Saarbruecken
Germany
* Other ftp sites are
ftp://crlftp.nmsu.edu/pub/non-lexical/NL_Software_Registy
ftp://dri.cornell.edu/pub/Natural_Language_Software_Registry
Part of Speech Tagger
* Description: A rule-based part of speech tagger developed by Eric
Brill.
* Availability: The tagger software, about 10 descriptive papers and
related data are available by anonymous ftp from
ftp://ftp.cs.jhu.edu/pub/brill/
___________________________________________________________________________
Copyright (c) 1993-6 by Andrew Hunt, all rights reserved.
This FAQ may be posted to any USENET newsgroup, on-line service, or BBS as
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This FAQ may not be distributed for financial gain.
This FAQ may not be included in any collections or compilations
without express permission from the author.
---
Andrew Hunt
Speech Applications Group
Sun Microsystems Laboratories Ph: (978) 442-2681
2 Elizabeth Drive, MS UCHL03-207 Fax: (978) 250-5067
Chelmsford, MA 01824, USA Email: andrew.hunt@east.sun.com
Last updated: Mon Jul 13 00:05:49 1998