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Topic: Speech encoding

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In the News (Wed 26 Jun 19)

  Kids.Net.Au - Encyclopedia > Speech coding
Speech coding is the compression of speech (into a code) for transmission using audio signal processing and speech processing techniques.
The techniques used in speech coding are similar to that in audio compression and audio coding[?] where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system.
In addition to the actual speech coding of the signal, it is often necessary to use channel coding for transmission, to avoid losses due to transmission errors.
www.kids.net.au /encyclopedia-wiki/sp/Speech_coding   (363 words)

 Speech encoding - Wikipedia, the free encyclopedia
Speech coding is the compression of speech (into a code) for transmission with speech codecs that use audio signal processing and speech processing techniques.
The techniques used in speech coding are similar to that in audio data compression and audio coding where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system.
In speech coding, the most important criterion is preservation of intelligibility and "pleasantness" of speech, with a constrained amount of transmitted data.
en.wikipedia.org /wiki/Speech_encoding   (455 words)

 Linear predictive speech encoding systems with efficient combination pitch coefficients computation - Patent 5794182   (Site not responding. Last check: 2007-11-03)
Further, speech processing is often utilized to generate or synthesize waveforms to resemble speech, to provide verification of a speaker's identity, and/or to translate speech inputs into written outputs.
In some speech processing systems, speech coding is performed to reduce the amount of data required for signal representation, often with analysis by synthesis adaptive predictive coders, including various versions of vector or code-excited coders.
Generation typically occurs with a speech signal being encoded, transmitting the codes for the signal, and decoding the codes to provide a decoded speech signal, which should be similar to the encoded speech signal.
www.freepatentsonline.com /5794182.html   (2974 words)

 Apparatus for preventing use of erroneous speech encoding parameters for decoding digital speech transmissions
A speech signal is transmitted over a radio path by a mobile station, and is received in a first base station.
The speech signal, encoded using encoding parameters, is forwarded in the information frames from the first base station to a remotely located speech decoder.
Speech encoding parameters are extracted from the received information frames, and the recurring synchronization check bit is monitored within the received information in order to detect an error in the recurring synchronization check bit in the received information frames.
www.patentalert.com /docs/000/z00005878.shtml   (312 words)

Levelt (1989) proposed that it is a phonetic representation, the output of phonetic encoding.
Speech errors also provide support for a morphological level of form encoding that is distinct from a lemma level with morphosyntactic parameters.
The issue for phonological encoding is what should be minimally specified in the mental lexicon for the speaker to build up, "from left to right", a metrically fully specified phonological word with complete specification of its phonological segments, their order, and syllabification.
www.bbsonline.org /Preprints/OldArchive/bbs.levelt.html   (20997 words)

 Encoding mode control method and decoding mode determining apparatus - Patent 6195337
Base station BS instructs the speech encoder to switch to the corresponding speech encoding mode and also instructs the channel encoder to switch to the corresponding channel encoding mode.
Speech encoder 30 may be a variable rate or embedded speech encoder (a variable rate encoder may also be implemented by switching between several fixed rate encoders).
Encoded speech is forwarded to a channel encoder 32, which has a channel encoding mode corresponding to each speech encoder mode.
www.freepatentsonline.com /6195337.html   (6122 words)

 Intelligent Speech Synthesis   (Site not responding. Last check: 2007-11-03)
In speech recognition the inclusion of linguistic and phonetic knowledge has come later, as the orthography to phoneme dictionary result of failure to find linguistic conversion units in the soundwave itself no matter how sophisticated the method of acoustic analysis adopted.
Although it may well seem obvious that speech would perhaps be more efficient without variability and although it may well be the case that the human perceptual system itself normalises the variability it hears, it is nevertheless the case that a human listener is extremely sensitive to the removal of variability from speech.
In speech synthesis and automatic speech recognition, however, the focus of attention is precisely on the encoding and decoding procedures.
www.essex.ac.uk /speech/archive/intell-ss/intell-ss.html   (5611 words)

The sentences are examples of our hybrid speech synthesis, in which only those select segments required to characterize a speaker are actually taken from that speaker's voice.
Our ability to create these illusions results from our team's extensive experience in knowledge-based speech synthesis and related technologies, which enables us to predict for any given language which aspects of a speech signal and which types of signal manipulations will be perceptible to the human ear and which will not.
Building on a powerful set of perceptual models, we are currently developing improved methods of speech synthesis and speech encoding, and are planning on extending our novel methods into other areas of speech processing.
www.novaspeech.com   (168 words)

 11 Transcriptions of Speech   (Site not responding. Last check: 2007-11-03)
Speech regarded as a purely acoustic phenomenon may well require different methods from those outlined here, as may speech regarded solely as a process of social interaction.
Speech varies according to a large number of dimensions, many of which have no counterpart in writing (for example, tempo, loudness, pitch, etc.).
Speech itself, though it may be represented as words, frequently contains items such as vocalized pauses which, although only semi-lexical, have immense importance in the analysis of spoken text.
www.tei-c.org /Vault/GL/P3/TS.htm   (7492 words)

 Northwestern University Interdepartmental Neuroscience Program
All work in the lab is motivated by interest in the linkages between neural encoding of speech and music in the central auditory nervous system, the resultant perception of those sounds, and subsequent incorporation of perception into higher-level language (e.g.
In a project called Listening, Learning and the Brain (http://www.communication.northwestern.edu/brainvolts/projects/listen/), neurophysiologic measures of speech encoding are related to speech-sound perception and to standardized measures of academic achievement in normal children and children with learning problems.
Biological encoding of speech and music sounds is often assumed to be primarily an auditory system process.
www.northwestern.edu /nuin/fac/kraus.htm   (755 words)

 Visible Speech / Physiological Alphabet
Visible Speech is a writing system invented in 1867 by Alexander Melville Bell, father of Alexander Graham Bell, the inventor of the telephone.
Visible Speech was also the first notation system for the sounds of speech independent of a particular language or dialect and was widely used to teaching students how to speak with a "standard" accent.
Visible Speech symbols are intended to provide visual representations of the positions the organs of speech need to be in to articulate individual sounds.
www.omniglot.com /writing/visiblespeech.htm   (137 words)

 Speech recognition using dynamic features (US5615299)
The class is an indication of the proximity of a speech segment to neighboring phones.
A feature vector is derived which represents the speech signal for a speech segment or series of speech segments for each frame.
This speech encoding technique is capable of being used in speech recognition systems by utilizing models, in which each model transition is tagged with one of the N classes.
www.delphion.com /details?pn=US05615299__   (597 words)

"The hypothesis that speech related movements are primarily a function of effortful, disrupted speech encoding is not supported by the available, but limited data, from aphaisa." pp.
For example, semiotically speaking, the link between the object of the speech and the object of the gesture must be established before we can look at timing and other factors; otherwise, one might do the same for gestural praxis and speech, which of course makes no sense since they are not semiotically related.
If gesture is not speech illustration and if it expresses the same information as speech but in a very different (holistic) way, then we need to ask why Japanese gesture is so different from English gesture.
www.unm.edu /~morford/Week13.html   (1434 words)

 TMH KTH :: Annual Report 1999
Applications of these methods to the coding of speech, audio, images, and image sequences, including the procedures found in the GSM, JPEG, and MPEG standards, are also described.
Some aspects of human communication by speech and hearing are also discussed in the framework of signal classification theory.
The Speech and Music Communication programme includes studies of human communication primarily with the help of acoustic signals such as speech and music.
www.speech.kth.se /annualreport/1999/ar-teaching.html   (1402 words)

 Simple Speech
This method is relatively economic in its use of memory for storage of the speech
No speech data will be stored in memory until the Acquisition Enable button is pressed.
be stored in its encoded form, in which case the acquisition procedure would not have to be repeated.
members.tripod.com /werdav/sspeech.html   (2142 words)

 Need help in encoding mono speech to AAC - Hydrogenaudio Forums
All speech that I record is mono 44.1kHz WAV format.
For reasonable to high quality speech I think a sample freq of 32kHz should be ok. And resampling to 22.05kHz seems not to be a bad idea for lower bitrates.
HE AAC mono speech encoded at 44kHz in Nero did help and I could get a nice sounding HE AAC file at 32kbps or 48kbps.
www.hydrogenaudio.org /forums/index.php?showtopic=18568   (642 words)

 SPIRIT in the Press   (Site not responding. Last check: 2007-11-03)
SPIRIT's ICELP speech codec algorithm is functioning on 10 KHz bandwidth that exceeds standard specifications for wideband applications and provides top speech quality required in multimedia applications and speech storage devices.
The multi-channel software codes speech signals and decodes 16-bit PCM bit stream at 20 KHz sampling rate, providing 32kbps compression rate.
Speech encoding/decoding is performed independently on the frame-by-frame basis, with frame length 16 ms and algorithmic delay 20 ms.
www.spiritdsp.com /press_30_10.html   (216 words)

 Spring Seminar on Models of Speech Production
In examining these models we will explore several issues that have captured the attention of the field (e.g., feed-forward vs. feedback approaches, levels or types of representation, etc.), as well as learn about various methods that are employed to examine these issues (e.g., analyses of errors, picture-naming tasks, eye-tracking, neuro-physiological techniques).
An event approach to the study of speech perception from a direct-realist perspective.
Watkins and Paus (2004) Modulation of motor excitability during speech perception: The role of broca's area.
people.ku.edu /~mvitevit/P800SpeechProd.html   (1015 words)

 DataCompression.info - Speech   (Site not responding. Last check: 2007-11-03)
Nellymoser is the leading provider of proprietary speech and audio software technology and solutions in the areas of compression, modification, synchronization and conversion.
These folks at UCSB are encoding speech at 2.8 Kbps.
This is supposed to be a 2.4 Kbps LPC speech codec.
datacompression.info /Speech.shtml   (2714 words)

 General Catalyst Partners :: News
Vocoding is used to dramatically reduce the data transmission rate for speech communications while maintaining acceptable levels of speech intelligibility and quality as well as good speaker recognition.
In the first phase of the Advanced Speech Encoding (ASE 1) program, BBN developed a conceptually new approach that met program goals of achieving noise robust vocoding that provided the same intelligibility in a harsh noise environment as current vocoders provide in low noise environments.
This approach was based on a new algorithmic method for analysis of speech signals in the presence of noise and on the exploitation of multiple sensors, including non-acoustic as well as acoustic sensors.
www.generalcatalyst.com /news/articles/bbn_050712.html   (496 words)

 Royalty Free Encoding/Decoding Library for Speech Compression, Decompression
The dsPIC30F Speech Encoding/Decoding Library is based on the Speex open source compression format, and it achieves a 16:1 compression ratio and toll quality speech (PESQ 3.7-4.2).
The speech encoder samples data at 8 kHz, using either the dsPIC DSC’s on-chip 12-bit ADC or a 16-bit input obtained from an external CODEC, via the on-chip codec interface port.
The library permits the compressed speech data to be stored in on-chip memory (Flash, EEPROM or RAM) or to an external Flash memory.
www.ferret.com.au /articles/1b/0c03981b.asp   (347 words)

 SMSW Web Page   (Site not responding. Last check: 2007-11-03)
However, even if speech processing is not modality-bound, the phonetic informational differences associated with modality can be expected to affect the efficiency of perceptual encoding of spoken words, as informational differences between natural and synthetic speech do (e.g., Pisoni and Greene, 1990).
The classical account of this immunity of retrieval operations from stimulus quality factors is that the operands of the retrieval process are represented in an abstract internal code from which stimulus quality properties have been removed (Sternberg, 1967).
Therefore, if it is appropriate to conceive of V speech stimuli simply as degraded (informationally poorer) forms of AV or A events, working memory representations arising from V speech signals should be retrieved with the same speed as representations arising from A or AV signals.
www.wramc.amedd.army.mil /departments/aasc/avlab/smswwebpage.htm   (1685 words)

 NIH Poster   (Site not responding. Last check: 2007-11-03)
Abnormally slow word recognition might be a factor in the speech understanding difficulties experienced by hearing-impaired patients, and might limit the benefit they derive from hearing aids.
The intercept of the linear model of a subject's memory set size-RT function is assumed to represent the sum of encoding speed and motor execution speed (Sternberg, 1975).
Apparently, in order to obtain AV encoding speed benefit, it is helpful for an individual's A and V unimodal encoding speeds to be similar.
www.wramc.amedd.army.mil /departments/aasc/avlab/nihwebpage.htm   (1783 words)

 Contents of Speech Input/Output
All HTK speech input is controlled by configuration parameters which give details of what processing operations to apply to each input speech file or audio source.
Also described are the facilities for augmenting the basic speech parameters with energy measures, delta coefficients and acceleration (delta-delta) coefficients and for splitting each parameter vector into multiple data streams to form observations.
The chapter concludes with a brief description of the tools HL and HC which are provided for viewing, manipulating and encoding speech files.
www.icsi.berkeley.edu /Speech/docs/HTKBook/node56_ct.html   (224 words)

 Net.Speech -- Desktop Audio comes to the Net   (Site not responding. Last check: 2007-11-03)
Both compress speech data so it can be transmitted and received by a 14.4 Kb/s modem, allowing people with dial-up IP accounts to use them.
For perspective, we should recall that early telephones produced sound at the speaker with about 1/10,000 the power of the original speech, even when the speaker and mike were directly connected [6].
For speech recording the quality seems comparable to that achieved with a full-size sound card and desktop mike, and the automatic gain control on the sound card makes for trouble free operation.
som.csudh.edu /cis/lpress/speech.htm   (4148 words)

 Royalty-free library for 16-bit DSC speech codec apps available
To boost wider use of its 16-bit digital signal controller in embedded audio applications, Microchip is making available a royalty-free speech encoding (compression) and decoding (decompression) library for its dsPIC family.
"Many design engineers are seeking to add speech compression to their bandwidth- or memory-constrained applications, but have opted for lower compression methodologies to avoid a stiff per-unit royalty for the intellectual property," said Mitra.
The speech encoder samples data at 8 kHz, using either the dsPIC DSC's on-chip 12-bit analog-to-digital converter or a 16-bit input obtained from an external CODEC, via the on-chip codec interface port.
www.embedded.com /shared/printableArticle.jhtml?articleID=164300413   (582 words)

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