Lectures on communication theory

"April 3, 1952." Lectures from the Research Laboratory of Electronics, Massachusetts Institute of Technology, Fall Term, 1951.Bibliography: p. 47-48.Army Signal Corps Contract No. DA36-039 sc-100 Project No. 8-102B. Dept. of the Army Project No. 3-99-10-022.D. Gabor

Simulation and implementation of a linear predictive coder

The main objective of this research was to design and build a Linear Predictive Coder (LPC) based on the TMS320 processor, and to incorporate this in the design of a low bit rate voice coding server for a Cambridge Ring. In order to decide on a suitable algorithm for the LPC, extensive simulations were carried out on a BBC computer. The computer used was interfaced to a frame store which, although its original purpose was to store video information, acted as a suitable store for speech. Up to six seconds of speech could be fed in from a microphone in real time for analysis. The BBC was fitted with a second processor, but in spite of this the processing times were very slow. [Continues.

Glottal-synchronous speech processing

Glottal-synchronous speech processing is a field of speech science where the pseudoperiodicity of voiced speech is exploited. Traditionally, speech processing involves segmenting and processing short speech frames of predefined length; this may fail to exploit the inherent periodic structure of voiced speech which glottal-synchronous speech frames have the potential to harness. Glottal-synchronous frames are often derived from the glottal closure instants (GCIs) and glottal opening instants (GOIs). The SIGMA algorithm was developed for the detection of GCIs and GOIs from the Electroglottograph signal with a measured accuracy of up to 99.59%. For GCI and GOI detection from speech signals, the YAGA algorithm provides a measured accuracy of up to 99.84%. Multichannel speech-based approaches are shown to be more robust to reverberation than single-channel algorithms. The GCIs are applied to real-world applications including speech dereverberation, where SNR is improved by up to 5 dB, and to prosodic manipulation where the importance of voicing detection in glottal-synchronous algorithms is demonstrated by subjective testing. The GCIs are further exploited in a new area of data-driven speech modelling, providing new insights into speech production and a set of tools to aid deployment into real-world applications. The technique is shown to be applicable in areas of speech coding, identification and artificial bandwidth extension of telephone speec

A Study of Accomodation of Prosodic and Temporal Features in Spoken Dialogues in View of Speech Technology Applications

Inter-speaker accommodation is a well-known property of human speech and human interaction in general. Broadly it refers to the behavioural patterns of two (or more) interactants and the effect of the (verbal and non-verbal) behaviour of each to that of the other(s). Implementation of thisbehavior in spoken dialogue systems is desirable as an improvement on the naturalness of humanmachine interaction. However, traditional qualitative descriptions of accommodation phenomena do not provide sufficient information for such an implementation. Therefore, a quantitativedescription of inter-speaker accommodation is required. This thesis proposes a methodology of monitoring accommodation during a human or humancomputer dialogue, which utilizes a moving average filter over sequential frames for each speaker. These frames are time-aligned across the speakers, hence the name Time Aligned Moving Average (TAMA). Analysis of spontaneous human dialogue recordings by means of the TAMA methodology reveals ubiquitous accommodation of prosodic features (pitch, intensity and speech rate) across interlocutors, and allows for statistical (time series) modeling of the behaviour, in a way which is meaningful for implementation in spoken dialogue system (SDS) environments.In addition, a novel dialogue representation is proposed that provides an additional point of view to that of TAMA in monitoring accommodation of temporal features (inter-speaker pause length and overlap frequency). This representation is a percentage turn distribution of individual speakercontributions in a dialogue frame which circumvents strict attribution of speaker-turns, by considering both interlocutors as synchronously active. Both TAMA and turn distribution metrics indicate that correlation of average pause length and overlap frequency between speakers can be attributed to accommodation (a debated issue), and point to possible improvements in SDS “turntaking” behaviour. Although the findings of the prosodic and temporal analyses can directly inform SDS implementations, further work is required in order to describe inter-speaker accommodation sufficiently, as well as to develop an adequate testing platform for evaluating the magnitude ofperceived improvement in human-machine interaction. Therefore, this thesis constitutes a first step towards a convincingly useful implementation of accommodation in spoken dialogue systems

Comparison of Wideband Earpiece Integrations in Mobile Phone

Perinteisesti puhelinverkoissa välitettävä puhe on ollut kapeakaistaista, kaistan ollessa 300 - 3400 Hz. Voidaan kuitenkin olettaa, että laajakaistaiset puhepalvelut tulevat saamaan markkinoilla enemmän jalansijaa tulevina vuosina. Tässä lopputyössä esitellään puheenkoodauksen perusteet laajakaistaisen adaptiivisen moninopeuspuhekoodekin (AMR-WB) kanssa. Laajakaistainen puhekoodekki laajentaa puhekaistan 50-7000 Hz käyttäen 16 kHz näytetaajuutta. Käytännössä laajempi kaista tarkoittaa parannuksia puheen ymmärrettävyyteen ja tekee siitä luonnollisemman ja mukavamman kuuloista. Tämän lopputyön päätavoite on vertailla kahden eri laajakaistaisen matkapuhelinkuulokkeen integrointia. Kysymys kuuluu, kuinka paljon käyttäjä hyötyy isommasta kuulokkeesta matkapuhelimessa? Kuulokkeiden suorituskyvyn selvittämiseksi niille tehtiin objektiivisia mittauksia vapaakentässä. Mittauksia tehtiin myös puhelimelle pää- ja torsosimulaattorissa (HATS) johdottamalla kuuloke suoraan vahvistimelle, sekä lisäksi puhelun ollessa aktiivisena GSM ja WCDMA verkoissa. Objektiiviset mittaukset osoittivat kahden eri integroinnin väliset erot kuulokkeiden taajuusvasteessa ja särössä erityisesti matalilla taajuuksilla. Lopuksi tehtiin kuuntelukoe tarkoituksena selvittää erottaako loppukäyttäjä pienemmän ja isomman kuulokkeen välistä eroa käyttäen kapeakaistaisia ja laajakaistaisia puhelinääninäytteitä. Kuuntelukokeen tuloksien pohjalta voidaan sanoa, että käyttäjä erottaa kahden eri integroinnin erot ja miespuhuja hyötyy naispuhujaa enemmän isommasta kuulokkeesta laajakaistaisella puhekoodekilla.The speech in telecommunication networks has been traditionally narrowband ranging from 300 Hz to 3400 Hz. It can be expected that wideband speech call services will increase their foothold in the markets during the coming years. In this thesis speech coding basics with adaptive multirate wideband (AMR-WB) are introduced. The wideband codec widens the speech band to new range from 50 Hz to 7000 Hz using 16 kHz sampling frequency. In practice the wider band means improvements to speech intelligibility and makes it more natural and comfortable to listen to. The main focus of this thesis work is to compare two different wideband earpiece integrations. The question is how much the end-user will benefit from using a larger earpiece in a mobile phone? To find out speaker performance, objective measurements in free field were done for the earpiece modules. Measurements were performed also for the phone on head and torso simulator (HATS) by wiring the earpieces directly to a power amplifier and with over the air on GSM and WCDMA networks. The results of objective measurements showed differences between the earpiece integrations especially on low frequencies in frequency response and distortion. Finally the subjective listening test is done for comparison to see if the end-user notices the difference between smaller and larger earpiece integrations using narrowband and wideband speech samples. Based on these subjective test results it can be said that the user can differentiate between two different integrations and that a male speaker benefits more from a larger earpiece than a female speaker