- Text Lab 1 3 9 – A Text Transformation Toolkit Powerpoint
- Text Lab 1 3 9 – A Text Transformation Toolkit Online
- 1Full system
- 1.1Multilingual
- 1.2Language specific
- 2Front end (NLP part)
- 2.1Front end inc G2P
- 2.2Text normalization
- 2.3Dictionary related tools
- 3Backend (Acoustic part)
- 3.2HMM based
- 3.3DNN based
- 3.4Wavenet based
- 4End-to-end (text to audio)
- 5Signal processing
- 5.1Vocoder, Glottal modelling
- 5.2Pitch extractor
- 5.3Sample modelling
- 5.4Toolkits
- 6Singing synthesizer
- 7Ebook reader
- 8Various tools
- 9Articulatory synthesizer
- 10API/Library
- 11Visualization & annotation tools
- 12Resources
- 12.1Dictionary
Full system
Multilingual
Question: 1 3 A C 1 4 1 5 And Consider The Matrix Transformation T(v) = Av. Which Of The Following Vectors Is In The Range Of T? Select All That Apply.
- For i = 1:3 subplot(3,1,i) plot(t(1:100),X(i,1:100)) title('Row ',num2str(i), ' in the Time Domain') end For algorithm performance purposes, fft allows you to pad the input with trailing zeros. In this case, pad each row of X with zeros so that the length of each row is the next higher power of 2 from the current length.
- Paste the PDSysConsoleLabtoolkit.toolkit file into this new toolkits directory. Open the.toolkit file in a text editor. Replace the text on line 3 with the path to the lab install directory. Complete the toolkit script and run the transform system from a System Console toolkit 1.
- C 1 stretches it; 0 1 compresses it We can stretch or compress it in the x-direction by multiplying x by a constant. C 1 compresses it; 0 1 stretches it; Note that (unlike for the y-direction), bigger values cause more compression. We can flip it upside down by multiplying the whole function by −1: g(x) = −(x 2).
- 75 synteny plots (Fig. Notably, the development of TBtools was highly collaborative and 76 greatly motivated by the true needs of wet-lab biologists. In the past five years, the tool has 77 attracted over 15,000 stable users. Many of these users actively provide informative feedback.
Festival
Festival offers a general framework for building speech synthesis systems as well as including examples of various modules. As a whole it offers full text to speech through a number APIs: from shell level, though a Scheme command interpreter, as a C++ library, from Java, and an Emacs interface. Festival is multi-lingual (currently English (British and American), and Spanish) though English is the most advanced. Tools and documentation for build new voices are available through Carnegie Mellon's FestVox project
- Last update: 2015/01/06
- Link: http://www.cstr.ed.ac.uk/downloads/festival/2.4/
- Reference:
FreeTTS
FreeTTS is a speech synthesis system written entirely in the JavaTM programming language. It isbased upon Flite: a small run-time speech synthesis engine developed at Carnegie MellonUniversity. Flite is derived from the Festival Speech Synthesis System from the University ofEdinburgh and the FestVox project from Carnegie Mellon University.
- Last update: 2009-03-09
- Link: http://freetts.sourceforge.net/docs/index.php
- Reference:
MBROLA
The aim of the MBROLA project, initiated by the TCTS Lab of the Faculté Polytechnique de Mons(Belgium), is to obtain a set of diphone-based speech synthesizers for as many languages aspossible, and provide them free for non-commercial applications.
- Last update:
- Link: http://tcts.fpms.ac.be/synthesis/mbrola.html
- Reference:
MARY
MARY is a multi-lingual (German, English, Tibetan) and multi-platform (Windows, Linux, MacOs X andSolaris) speech synthesis system. It comes with an easy-to-use installer - no technical expertiseshould be required for installation. It enables expressive speech synthesis, using both diphone andunit-selection synthesis.
- Last update: 2017/09/26
- Link: http://mary.dfki.de/
- Reference:
AhoTTS
Text-to-Speech conversor for Basque, Spanish, Catalan, Galician and English.It includes linguistic processing and built voices for all the languages aforementioned. Its acoustic engine is based on htsengine and it uses a high quality vocoder called AhoCoder.
- Last update: 2015/07/15
- Link: https://sourceforge.net/projects/ahottsmultiling/
Language specific
AHOTTS (Basque & spanish)
Text-to-Speech conversor for Basque and Spanish. It includeslinguistic processing and built voices for the languagesaforementioned. Its acoustic engine is based on htsengine and it usesa high quality vocoder called AhoCoder. Receipts 1 8 1.
- Last update: 2016/04/07
- Link: https://sourceforge.net/projects/ahotts
- Link2: https://sourceforge.net/projects/ahottsiparrahotsa/ (for Lapurdian dialect of Basque.)
- Reference:
RHVoice (Russian)
RHVoice is a free and open source speech synthesizer.
- Last update: 2017/09/24
- Link: https://github.com/Olga-Yakovleva/RHVoice
Front end (NLP part)
Front end inc G2P
SiRE
(Si)mply a (Re)search front-end for Text-To-Speech Synthesis.This is a research front-end for TTS. It is incomplete, inconsistent, badly coded and slow.But it is useful for me and should slowly develop into something useful to others.
- Last update: 2016/10/11
- Link: https://github.com/RasmusD/SiRe
Phonetisaurus
This repository contains scripts suitable for training, evaluating and using grapheme-to-phoneme models for speech recognition using the OpenFst framework. The current build requires OpenFst version 1.6.0 or later, and the examples below use version 1.6.2.
The repository includes C++ binaries suitable for training, compiling, and evaluating G2P models. It also some simple python bindings which may be used to extract individual multigram scores, alignments, and to dump the raw lattices in .fst format for each word.
- Last update: 2017/09/17
- Link: https://github.com/AdolfVonKleist/Phonetisaurus
Ossian
Ossian is a collection of Python code for building text-to-speech (TTS) systems, with an emphasis on easing research into building TTS systems with minimal expert supervision. Work on it started with funding from the EU FP7 Project Simple4All, and this repository contains a version which is considerable more up-to-date than that previously available. In particular, the original version of the toolkit relied on HTS to perform acoustic modelling. Although it is still possible to use HTS, it now supports the use of neural nets trained with the Merlin toolkit as duration and acoustic models. All comments and feedback about ways to improve it are very welcome.
- Last update: 2017/09/15
- Link: https://github.com/CSTR-Edinburgh/Ossian
SALB
The SALB system is a software framework for speech synthesis using HMM based voice models built by HTS (http://hts.sp.nitech.ac.jp/). See a more generic description on http://m-toman.github.io/SALB/.
The package currently includes:
A C++ framework that abstracts the backend functionality and provides a SAPI5 interface, a command line interface and a C++ API.
Backend functionality is provided by
- an internal text analysis module for (Austrian) German,
- flite as text analysis module for English and
- htsengine for parameter generation/synthesis. (see COPYING for information on 3rd party libraries)
Also included is an Austrian German male voice model.
- Last update: 2016/11/14
- Link: https://github.com/m-toman/SALB
Sequence-to-Sequence G2P toolkit
The tool does Grapheme-to-Phoneme (G2P) conversion using recurrent neural network (RNN) with long short-term memory units (LSTM). LSTM sequence-to-sequence models were successfully applied in various tasks, including machine translation [1] and grapheme-to-phoneme [2].
This implementation is based on python TensorFlow, which allows an efficient training on both CPU and GPU.
- Last update: 2017/03/28
- Link: https://github.com/cmusphinx/g2p-seq2seq
Text normalization
Sparrowhawk
Sparrowhawk is an open-source implementation of Google's Kestrel text-to-speechtext normalization system. It follows the discussion of the Kestrel system asdescribed in:
Ebden, Peter and Sproat, Richard. 2015. The Kestrel TTS text normalizationsystem. Natural Language Engineering, Issue 03, pp 333-353.
After sentence segmentation (sentenceboundary.h), the individual sentences arefirst tokenized with each token being classified, and then passed to thenormalizer. The system can output as an unannotated string of words, and richerannotation with links between input tokens, their input string positions, andthe output words is also available.
- Last update: 2017/07/25
- Link: https://github.com/google/sparrowhawk
ASRT
This is the README for the Automatic Speech Recognition Tools.
This project contains various scripts in order to facilitate the preparation of ASR related tasks.
Current tasks ares:
- Sentences extraction from pdf files
- Sentences classification by langues
- Sentences filtering and cleaning
Document sentences can be extracted into single document or batch mode.
For an example on how to extract sentences in batch mode, please have a look at the rundatapreparationtask.sh script located in examples/bash directory.
For an example on how to extract sentences in single document mode, please have a look at the rundatapreparation.sh script located in examples/bash directory.
The is also an API to be used in python code. It is located into the common package and is called DataPreparationAPI.py
- Last update: 2017/09/20
- Link: https://github.com/idiap/asrt
IRISA text normalizer
Text normalisation tools from IRISA lab.
The tools provided here are split into 3 steps:
- Tokenisation (adding blanks around punctation marks, dealing with special cases like URLs, etc.)
- Generic normalization (leading to homogeneous texts where (almost) information have been lost and where tags have been added for some entities)
- Specific normalisation (projection of the generic texts into specific forms)
- Last update: 2018/01/09
- Link: https://github.com/glecorve/irisa-text-normalizer
Dictionary related tools
CMU Pronunciation Dictionary Tools
Tools for working with the CMU Pronunciation Dictionary
- Last update: 2015/02/23
- Link: https://github.com/cmusphinx/cmudict-tools
ISS scripts for dictionary maintenance
These scripts are sufficient to convert the distributed forms of dictionaries into forms useful for our tools (notably HTK and ISS). Once a dictionary is in a standard form, the generic tools in ISS can be used to manipulate it further.
- Last update: 2017/07/04
- Link: https://github.com/idiap/iss-dicts
Backend (Acoustic part)
Unit selection
HMM based
MAGE
MAGE is a C/C++ software toolkit for reactive implementation of HMM-based speech and singing synthesis.
- Last update: 2014/07/18
- Link: https://github.com/numediart/mage
HMM-Based Speech Synthesis System (HTS)
The basic core system of HTS, available from NITECH, was implemented as a modified version of HTKtogether with SPTK (see below), and is released as HMM-Based Speech Synthesis System (HTS) in a formof patch code to HTK.
- Last update: 2016/12/25
- Link: http://hts.sp.nitech.ac.jp/
HTS Engine
htsengine is a small run-time synthesis engine (less than 1 MB including acoustic models), whichcan run without the HTK library. The current version does not include any text analyzer but theFestival Speech Synthesis System can be used as a text analyzer.
- Last update: 2015/12/25
- Link: http://hts-engine.sourceforge.net/
DNN based
MERLIN
Merlin is a toolkit for building Deep Neural Network models for statistical parametric speech synthesis. It must be used in combination with a front-end text processor (e.g., Festival) and a vocoder (e.g., STRAIGHT or WORLD).
The system is written in Python and relies on the Theano numerical computation library.
Merlin comes with recipes (in the spirit of the Kaldi automatic speech recognition toolkit) to show you how to build state-of-the art systems.
- Last update: 2017/09/29
- Link: http://www.cstr.ed.ac.uk/projects/merlin
- Reference:
IDLAK
Idlak is a project to build an end-to-end parametric TTSsystem within Kaldi, to be distributed with the same licence.
It contains a robust front-end, voice building tools, speech analysisutilities, and DNN tools suitable for parametric synthesis. It also containsan example of using Idlak as an end-to-end TTS system, in egs/ttsdnnarctic/s1
Note that the kaldi structure has been maintained and the tool buildingprocedure is identical.
- Last update: 2017/07/03
- Link: https://github.com/bpotard/idlak
- Reference:
CURRENNT scripts
The scripts and examples on the modified CURRENNT toolkit
- Last update: 2017/08/27
- Link: https://github.com/TonyWangX/CURRENNT_SCRIPTS
Wavenet based
tensorflow-wavenet
A TensorFlow implementation of DeepMind's WaveNet paper
- Last update: 2017/05/23
- Link: https://github.com/ibab/tensorflow-wavenet
Other
End-to-end (text to audio)
barronalex/Tacotron
Implementation of Google's Tacotron in TensorFlow
- Last update: 2017/08/08
- Link: https://github.com/barronalex/Tacotron
keithito/tacotron
A TensorFlow implementation of Google's Tacotron speech synthesis with pre-trained model
- Last update: 2017/11/06
- Link: https://github.com/keithito/tacotron
Char2Wav: End-to-End Speech Synthesis
This repo has the code for our ICLR submission:
Jose Sotelo, Soroush Mehri, Kundan Kumar, João Felipe Santos, Kyle Kastner, Aaron Courville, Yoshua Bengio. Char2Wav: End-to-End Speech Synthesis.
The website is here.
- Last update: 2017/02/28
- Link: https://github.com/sotelo/parrot
- Reference:
Signal processing
Vocoder, Glottal modelling
STRAIGHT
STRAIGHT is a tool for manipulating voice quality, timbre, pitch, speed and other attributesflexibly. It is an always evolving system for attaining better sound quality, that is close to theoriginal natural speech, by introducing advanced signal processing algorithms and findings incomputational aspects of auditory processing.
STRAIGHT decomposes sounds into source information and resonator (filter) information. Thisconceptually simple decomposition makes it easy to conduct experiments on speech perception usingSTRAIGHT, the initial design objective of this tool, and to interpret experimental results in termsof huge body of classical studies.
- Last update:
- Link: http://www.wakayama-u.ac.jp/~kawahara/STRAIGHTadv/index_e.html
- Reference:
World
WORLD is free software for high-quality speech analysis, manipulation and synthesis. It can estimate Fundamental frequency (F0), aperiodicity and spectral envelope and also generate the speech like input speech with only estimated parameters.
This source code is released under the modified-BSD license. There is no patent in all algorithms in WORLD.
- Last update: 2017/08/23
- Link: https://github.com/mmorise/World
- Reference:
Covarep - A Cooperative Voice Analysis Repository for Speech Technologies
Covarep is an open-source repository of advanced speech processing algorithmsand is stored as a GitHub project (https://github.com/covarep/covarep) whereresearchers in speech processing can store original implementations of publishedalgorithms.
Over the past few decades a vast array of advanced speech processing algorithmshave been developed, often offering significant improvements over the existingstate-of-the-art. Such algorithms can have a reasonably high degree ofcomplexity and, hence, can be difficult to accurately re-implement based onarticle descriptions. Another issue is the so-called 'bug magnet effect' withre-implementations frequently having significant differences from the originalones. The consequence of all this has been that many promising developmentshave been under-exploited or discarded, with researchers tending to stick toconventional analysis methods.
By developing Covarep we are hoping to address this by encouraging authors toinclude original implementations of their algorithms, thus resulting in asingle de facto version for the speech community to refer to.
- Last update: 2016/10/16
- Link: https://github.com/covarep/covarep
- Reference:
MagPhase Vocoder
Speech analysis/synthesis system for TTS and related applications.
This software is based on the method described in the paper:
- Espic, C. Valentini-Botinhao, and S. King, “Direct Modelling of Magnitude and Phase Spectra for Statistical Parametric Speech Synthesis,” in Proc. Interspeech, Stockholm, Sweden, August, 2017.
- Last update: 2017/08/30
- Link: https://github.com/CSTR-Edinburgh/magphase
- Reference:
WavGenSR
Waveform generator based on signal reshaping for statistical parametric speech synthesis.
- Last update: 2017/08/30
- Link: https://github.com/CSTR-Edinburgh/WavGenSR
- Reference:
Pulse model analysis and synthesis
It is basically the vocoder described in:
- Degottex, P. Lanchantin, and M. Gales, 'A Pulse Model in Log-domain for a Uniform Synthesizer,' in Proc. 9th Speech Synthesis Workshop (SSW9), 2016.
- Last update: 2017/09/7
- Link: https://github.com/gillesdegottex/pulsemodel
- Reference:
YANG VOCODER: Yet-ANother-Generalized VOCODER
Yet another vocoder that is not STRAIGHT.
This project is a state-of-the-art vocoder that parameterizes the speech signalinto a parameterization that is amenable to statistical manipulation.
The VOCODER was developed by Hideki Kawahara during his internship at Google.
- Last update: 2017/01/02
- Link: https://github.com/google/yang_vocoder
Ahocoder
Ahocoder parameterizes speech waveforms into three different streams: log-f0, cepstral representation of the spectral envelope, and maximum voiced frequency. It provides high accuracy during analysis and high quality during reconstruction. It is adequate for statistical parametric speech synthesis and voice conversion. Furthermore, it can be used just for basic speech manipulation and transformation (pitch level and variance, speaking rate, vocal tract length…).
Ahocoder is reported to be a very good complement for HTS. The output files generated by Ahocoder contain float numbers without header, so they are fully compatible with the HTS demo scripts in the HTS website. You can use the same configuration as in the STRAIGHT-based demo, using the 'bap' stream to handle maximum voiced frequency (set its dimension to 1 both in data/Makefile and in scripts/Config.pm).
- Last update: 2014
- Link: http://aholab.ehu.es/ahocoder/
PhonVoc: Phonetic and Phonological vocoding
This is a computational platform for Phonetic and Phonologicalvocoding, released under the BSD licence. See file COPYING fordetails. The software is based on Kaldi (v. 489a1f5) and Idiap SSP.For training of the analysis and synthesis models, follow pleasetrain/README.txt.
- Last update: 2016/11/23
- Link: https://github.com/idiap/phonvoc
GlottGAN
Generative adversarial network-based glottal waveform model for statistical parametric speech synthesis
- Last update: 2017/05/30
- Link: https://github.com/bajibabu/GlottGAN
- Reference:
Postfilt gan
This is an implementation of 'Generative adversarial network-based postfilter for statistical parametric speech synthesis'
Please check the run.sh file to train the system. Currently, testing part is not yet implemented.
- Last update: 2017/07/06
- Link: https://github.com/bajibabu/postfilt_gna
- Reference:
Pitch extractor
REAPER: Robust Epoch And Pitch EstimatoR
This is a speech processing system. The reaper program uses the EpochTracker class to simultaneously estimate the location of voiced-speech 'epochs' or glottal closure instants (GCI), voicing state (voiced or unvoiced) and fundamental frequency (F0 or 'pitch'). We define the local (instantaneous) F0 as the inverse of the time between successive GCI.
This code was developed by David Talkin at Google. This is not an official Google product (experimental or otherwise), it is just code that happens to be owned by Google.
- Last update: 2015/03/04
- Link: https://github.com/google/REAPER
SSP - Speech Signal Processing module
SSP is a package for doing signal processing in python; the functionality is biassed towards speech signals. Top level programs include a feature extracter for speech recognition, and a vocoder for both coding and speech synthesis. The vocoder is based on linear prediction, but with several experimental excitation models. A continuous pitch extraction algorithm is also provided, built around standard components and a Kalman filter.
There is a 'sister' package, libssp, that includes translations of some algorithms in C++. Libssp is built around libube that makes this translation easier.
SSP is released under a BSD licence. See the file COPYING for details.
- Last update: 2017/04/16
- Link: https://github.com/idiap/ssp
Sample modelling
SampleRNN
SampleRNN: An Unconditional End-to-End Neural Audio Generation Mode
- Last update:
- Link: https://github.com/soroushmehr/sampleRNN_ICLR2017
Toolkits
SPTK - Speech Signal Processing Toolkit
The main feature of the Speech Signal Processing Toolkit, available from NITECH, is that not onlystandard speech analysis and synthesis techniques (e.g., LPC analysis, PARCOR analysis, LSPanalysis, PARCOR synthesis filter, LSP synthesis filter, and vector quantization techniques) butalso speech analysis and synthesis techniques developed at the research group can easily be used.
- Last update: 2016/12/25
- Link: http://sp-tk.sourceforge.net/
Text Lab 1 3 9 – A Text Transformation Toolkit Powerpoint
Singing synthesizer
Sinsy
Sinsy is a HMM-based singing voice synthesis system.
- Last update: 2015/12/25
- Link: http://sinsy.sourceforge.net/
Ebook reader
Bard Storyteller ebook reader
Bard Storyteller is a text reader. Bard not only allows a user to read books, but can also read books to the user using text-to-speech. It supports txt, epub and (x)html files.
- Last update: 2014/07
Text Lab 1 3 9 – A Text Transformation Toolkit Online
- Link: http://festvox.org/bard/
Various tools
SparkNG
Matlab realtime speech tools and voice production tools
- Last update: 2017/06/29
- Link: http://www.wakayama-u.ac.jp/~kawahara/MatlabRealtimeSpeechTools/
Articulatory synthesizer
KLAIR - A virtual infant for spoken language acquisition research
The KLAIR project aims to build and develop a computational platform to assist research into the acquisition of spoken language. The main part of KLAIR is a sensori-motor server that displays a virtual infant on screen that can see, hear and speak. Behind the scenes, the server can talk to one or more client applications. Each client can monitor the audio visual input to the server and can send articulatory gestures to the head for it to speak through an articulatory synthesizer. Clients can also control the position of the head and the eyes as well as setting facial expressions. By encapsulating the real-time complexities of audio and video processing within a server that will run on a modern PC, we hope that KLAIR will encourage and facilitate more experimental research into spoken language acquisition through interaction.
- Last update:
- Link: http://www.phon.ucl.ac.uk/project/klair/
- Reference:
Vocaltractlab
VocalTractLab stands for 'Vocal Tract Laboratory' and is an interactive multimedial software tool to demonstrate the mechanism of speech production. It is meant to facilitate an intuitive understanding of speech production for students of phonetics and related disciplines.
The current versions of VocalTractLab are free of charge. Only a registration code, which you can request by email, will be necessary to activate the software. VocalTractLab is written for Windows operating systems (XP or higher), but a porting to Linux/Unix is conceivable for the future.
- Last update: 2016
- Link: http://www.vocaltractlab.de/
API/Library
Speech Tools
The Edinburgh Speech Tools Library is a collection of C++ class,functions and related programs for manipulating the sorts of objectsused in speech processing. It includes support for reading and writingwaveforms, parameter files (LPC, Ceptra, F0) in various formats andconverting between them. It also includes support for linguistic typeobjects and support for various label files and ngrams (withsmoothing).
In addition to the library a number of programs are included. Anintonation library which includes a pitch tracker, smoother andlabelling system (using the Tilt Labelling system), a classificationand regression tree (CART) building program called wagon. Also thereis growing support for various speech recognition classes such asdecoders and HMMs.
The Edinburgh Speech Tools Library is not an end in itself butdesigned to make the construction of other speech systems easy. It isfor example to provided the underlying classes in the Festival SpeechSynthesis System
The speech tools are currently distributed in full source form freefor unrestricted use.
- Last update: 2015/01/06
- Link: http://www.cstr.ed.ac.uk/projects/speech_tools/
ROOTS
Roots is an open source toolkit dedicated to annotated sequential data generation, management andprocessing. It is made of a core library and of a collection of utility scripts. A rich API isavailable in C++ and in Perl.
- Last update: 2015/07/01
- Link: http://roots-toolkit.gforge.inria.fr/
- Reference:
Visualization & annotation tools
Praat
Praat is a system for doing phonetics by computer. The computer program Praat is a research,publication, and productivity tool for phoneticians. With it, you can analyse, synthesize, andmanipulate speech, and create high-quality pictures for your articles and thesis.
- Last update:
- Link: http://www.fon.hum.uva.nl/praat/
- Reference:
KPE
KPE provides a graphical interface for the implementation of the Klatt 1980 formant synthesiser. Theinterface allows users to display and edit Klatt parameters using a graphical display which includesthe time-amplitude waveform of both the original speech and its synthetic copy, and some signalanalysis facilities.
- Last update:
- Link: http://www.speech.cs.cmu.edu/comp.speech/Section5/Synth/klatt.kpe80.html
Wavesurfer
WaveSurfer is a tool for doing speech analysis. The analysis features include formants and pitchextraction and real time spectrograms. The Wavesurfer tool built on top of the Snack speechvisualization module, is highly modular and extensible at several levels.
- Last update:
- Link: https://sourceforge.net/projects/wavesurfer/
Resources
Dictionary
Unisyn lexicon
The Unisyn lexicon is a master lexicon transcribed in keysymbols, a kind of metaphoneme which allows the encoding of multiple accents of English.
The lexicon is accompanied by a number of perl scripts which transform the base lexicon via phonological and allophonic rules, and other symbol changes, to produce output transcriptions in different accents. The rules can be applied to the whole lexicon, to produce an accent-specific lexicon, or to running text. Output can be displayed in keysymbols, SAMPA, or IPA.
The system uses a geographically-based accent hierarchy, with a tree structure describing countries, regions, towns and speakers; this hierarchy is used to specify the application of rules and other pronunciation features.
The lexicon system is customisable, and the documentation explains how to modify output by swtiching rules on and off, adding new rules or editing existing ones. The user can also add new nodes in the accent hierarchy (new accents or new speakers within an accent), or add new symbols.
A number of UK, US, Australian and New Zealand accents are included in the release.
The scripts run under unix, or Windows 98 (DOS), and use perl 5.6.0.
- Last update:
- Link: http://www.cstr.ed.ac.uk/projects/unisyn/
Combilex
Combilex GA is a keyword-based lexicon for the General American pronunciation.
The combilex contains c.145,000 entries, including the 20,000 most frequent words and contains a variety of linguistic information alongside detailed pronunciations, including many useful proper names.
Combilex GA is an ASCII text file, one entry-per-line, which is easily adaptable for use in text-to-speech synthesis (voice-building or run-time synthesis) and in speech recognition systems.
Full manually notated orthographic-phonemic correspondences are included, allowing derivation of accurate grapheme-to-phoneme rules.
- Last update:
- Link: https://licensing.edinburgh-innovations.ed.ac.uk/item.php?item=combilex-ga
- Reference:
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HAMLET. Are you fair?
Exactscan pro 19 1 – powerful fast document scanning software. OPHELIA. What means your lordship?
HAMLET. That if you be honest and fair, your honesty should admit no discourse to your beauty.
OPHELIA. Could beauty, my lord, have better commerce than with honesty?
HAMLET. Ay, truly; for the power of beauty will sooner transform honesty from what it is to a bawd than the force of honesty can translate beauty into his likeness. This was sometime a paradox, but now the time gives it proof. I did love you once.
OPHELIA. Indeed, my lord, you made me believe so.
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