A Study on Speaker Normalization Using Vocal Tract Normalization and Speaker Adaptive Training - Welling, L.; Haeb-Umbach, Reinhold; Aubert, X.; Haberland, N. (ICASSP 1998, Seattle)
Although speaker normalization is attempted in very different manners, vocal tract normalization (VTN) and speaker adaptive training (SAT) share many common properties. We show that both lead to more compact representations of the phonetically relevant variations of the training data and that both achieve improved error rate performance only if a complementary normalization or adaptation operation is conducted on the test data. Algorithms for fast test speaker enrollment are presented for both normalization methods: in the framework of SAT, a pre-transformation step is proposed, which alone, i.e. without subsequent unsupervised MLLR adaption, reduces the error rate by almost 10% on the WSJ 5k test sets. For VTN, the use of a Gaussian mixture model makes obsolete a first recognition pass to obtain a preliminary transcription of the test utterance at hardly and loss in performance.

Signal Representations for Hidden Markov Model Based On-Line Handwriting Recognition - Dolfing, J.G.A.; Haeb-Umbach, Reinhold (ICASSP, Munich)
Addresses the problem of online, writer-independent, unconstrained handwriting recognition. Based on hidden Markov models (HMM), which are successfully employed in speech recognition tasks, we focus on representations which address scalability, recognition performance and compactness. 'Delayed' features are introduced which integrate more global, handwriting specific knowledge into the HMM representation. These features lead to larger error-rate reduction than 'delta' features which are known from speech recognition and even require fewer additional components. Scalability is addressed with a size-independent representation. Compactness is achieved with linear discriminant analysis. The representations are discussed and the results for a mixed-style word recognition task with vocabularies of 200 (up to 99% correct words) and 20000 words (up to 88.8% correct words) are given.

The development of a command-based speech interface for a telephone answering machine - Gamm, Stephan; Haeb-Umbach, Reinhold; Langmann, Detlev (Speech Communication)
This paper reports the design of a command-based speech interface for an answering machine or a voice mail system. Automatic speech recognition was integrated in order to facilitate the remote control and the retrieval of voice messages from any telephone in a speech-only dialogue. The design goal was that consumers would perceive the speech interface as a benefit compared with the common touch-tone interface. In this paper we will first describe the speech technology underlying the system. Then it will be shown how, based on this technology, the user interface was designed in a top-down approach. We started with the development of a concept and tested it by means of a Wizard-of-Oz simulation. After refining the concept in parallel design, it was implemented in a high-fidelity prototype. By means of qualitative user testing the design was improved in three iteration steps. The achievement of the design goal was finally verified with user tests in two countries.

Robust Speech Recognition for Wireless Networks and Mobile Telephony - Haeb-Umbach, Reinhold (Eurospeech)
The increased popularity of mobile telephony introduces both challenges and opportunitites for automatic speech recognition. ASR offers ways to simplify the use of mobile phones, notably in hands- and eyes-busy situations. However, the acoustic environment can be severely degraded and the wireless network may add additional distortions to the speech signal. This paper gives an overview of the sources of degradation and attempts to robust speech recognition for mobile communications. Emphasis is placed on approaches which are suitable for implementation in mobile terminals. Two example applications are described which illustrate the robustness issues and design considerations typical of low-cost noisy speech recognition: voice-dialling in a GSM phone and hands-free digit recognition in the car.

European Speech Databases for Telephone Applications - Hoege, H.; Tropf, H. S.; Winsky, R.; van den Heuvel, H.; Haeb-Umbach, Reinhold; Choukri, K. (ICASSP, Munich)
The SpeechDat project aims to produce speech databases for all official languages of the European Union and some major dialectal variants and minority languages resulting in 28 speech databases. They will be recorded over fixed and mobile telephone networks. This will provide a realistic basis for training and assessment of both isolated and continuous-speech utterances, employing whole-word or subword approaches, and thus can be used for developing voice driven teleservices including speaker verification. The specification of the databases has been developed jointly, and is essentially the same for each language to facilitate dissemination and use. There will be a controlled variation among the speakers concerning sex, age, dialect, environment of call, etc. The validation of all databases will be carried out centrally. The SpeechDat databases will be transferred to ELRA for distribution. The next databases to be recorded will cover East European languages.

Acoustic Front Ends for Speaker-Independent Digit Recognition in Car Environments - Langmann, Detlev; Fischer, Alexander; Wuppermann, Friedhelm; Haeb-Umbach, Reinhold; Eisele, Thomas (Eurospeech)
This paper describes speaker-independent speech recognition experiments concerning acoustic front end processing on a speech database that was recorded in 3 different cars. We investigate different feature analysis approaches (mel-filter bank, mel-cepstrum, perceptually linear predictive coding) and present results with noise compensation techniques based on spectral subtraction. Although the methods employed lead to considerable error rate reduction the error analysis shows that low signal-to-noise ratios are still a problem

Investigation of Acoustic Front Ends for Speaker-Independent Speech Recognition in the Car - Langmann, Detlev; Wuppermann, Friedhelm; Haeb-Umbach, Reinhold; Fischer, A.; Eisele, Thomas (Aachener Kolloquium on Signal Theory)

A Comparative Study of Linear Feature Transformation Techniques for Automatic Speech Recognition - Eisele, Thomas; Haeb-Umbach, Reinhold; Langmann, Detlev (ICSLP , Philadelphia)
Although widely used, there are still open questions concerning which properties of linear discriminant analysis (LDA) account for its success in many speech recognition systems. In order to gain more insight into the nature of the transformation we compare LDA with mel-cepstral feature vectors with respect to the following criteria: decorrelation and ordering property; invariance under linear transforms; automatic learning of dynamical features; and data dependence of the transformation.

Findings with the Design of a Command-Based Speech Interface for a Voice Mail System - Gamm, Stephan; Haeb-Umbach, Reinhold; Langmann, Detlev (IEEE Workshop on Interactive Voice Technology for Telecommunications Applications)
This paper tells the story of the design of a command-based speech interface for a voice mail system. Speech recognition was integrated in the voice mail system in order to allow the remote interrogation of messages in a speech-only dialogue. Our design goal was that consumers would perceive voice control as a clear benefit versus touch-tone control. It is shown how the speech interface was designed in a top-down approach. We started with a concept development and tested it by means of a Wizard-of-Oz simulation. After refining the concept in parallel design, the design was implemented in a high-fidelity prototype. By means of qualitative user testing it was improved in three iteration steps. We verified the achievement of our design goal with tests in two countries

FRESCO: The French Telephone Speech Data Collection - Part of the European SpeechDat(M) Project - Langmann, Detlev; Haeb-Umbach, Reinhold (ICSLP, Philadelphia)
The paper describes the design, collection and postprocessing of the French SpeechDat corpus FRESCO. Being a database of approximately 35000 utterances recorded from 1000 callers over the terrestrial telephone network in France, it comprises immediately usable and relevant speech for the initial training and assessment of speaker independent phoneme model or word model based speech recognizers, as they are employed in automated telephone services. FRESCO is one of the 1000 speaker telephone speech databases produced as "case studies" within the European project SpeechDat(M).

Robust Rejection Modeling for a Small-Vocabulary Application - Langmann, Detlev; Haeb-Umbach, Reinhold; Eisele, Thomas (ITG Fachtagung Sprachkommunikation, Frankfurt)

Application of Clustering Techniques to Mixture Density Modelling for Continuous-Speech Recognition - Dugast, Christian; Beyerlein, Peter; Haeb-Umbach, Reinhold (ICASSP, Detroit)
Clustering techniques have been integrated at different levels into the training procedure of a continuous-density hidden Markov model (HMM) speech recognizer. These clustering techniques can be used in two ways. First acoustically similar states are tied together. It will help to reduce the number of parameters but also allow to train otherwise rarely seen states together with more robust ones (state-tying). Secondly densities are clustered across states, this reduces the number of densities while at the same time keeping the best performances of our recognizer (density-clustering). We have applied these techniques both to word-based small-vocabulary and phoneme-based large-vocabulary recognition tasks. On the WSJ task, we could achieve a reduction of the word error rate by 7%. On the TI/NIST-connected digit task, the number of parameters was reduced by a factor 2-3 while keeping the same string error rate.

User interface design of voice controlled consumer electronics - Gamm, Stephan; Haeb-Umbach, Reinhold (Philips Journal of Research)
Today speech recognition of a small vocabulary can be realized so cost-effectively that the technology can penetrate into consumer electronics. But, as first applications that failed on the market show, it is by no means obvious how to incorporate voice control in a user interface. This paper addresses the issue of how to design a voice control so that the user perceives it as a benefit. User interface guidelines that are adapted or specific to voice control are presented. Then the process of designing a voice control in the user-centred approach is described. By means of two examples, the car stereo and telephone answering machine, it is shown how this is turned into practice.

Human Factors of a Voice-Controlled Car Stereo - Gamm, Stephan; Haeb-Umbach, Reinhold (Eurospeech, Madrid)

The Usability Engineering of a Voice-Controlled Answering Machine - Gamm, Stephan; Haeb-Umbach, Reinhold; Langmann, Det (International Symposium on Human Factors in Telecommunications, Melbourne)

Speech recognition algorithms for voice control interfaces - Haeb-Umbach, Reinhold; Beyerlein, Peter; Geller, Dieter (Philips Journal of Research)
Recognition accuracy has been the primary objective of most speech recognition research, and impressive results have been obtained, e.g. less than 0.3% word error rate on a speaker-independent digit recognition task. When it comes to real-world applications, robustness and real-time response might be more important issues. For the first requirement we review some of the work on robustness and discuss one specific technique, spectral normalization, in more detail. The requirement of real-time response has to be considered in the light of the limited hardware resources in voice control applications, which are due to the tight cost constraints. In this paper we discuss in detail one specific means to reduce the processing and memory demands: a clustering technique applied at various levels within the acoustic modelling.

Automatic Transcription of Unknown Words in a Speech Recognition System - Haeb-Umbach, Reinhold; Beyerlein, P.; Thelen, E. (ICASSP, Detroit)
We address the problem of automatically finding an acoustic representation (i.e. a transcription) of unknown words as a sequence of subword units, given a few sample utterances of the unknown words, and an inventory of speaker-independent subword units. The problem arises if a user wants to add his own vocabulary to a speaker-independent recognition system simply by speaking the words a few times. Two methods are investigated which are both based on a maximum-likelihood formulation of the problem. The experimental results show that both automatic transcription methods provide a good estimate of the acoustic models of unknown words. The recognition error rates obtained with such models in a speaker-independent recognition task are clearly better than those resulting from separate whole-word models. They are comparable with the performance of transcriptions drawn from a dictionary.

The Philips Research system for continuous-speech dictation - Steinbiss, Volker; Ney, Hermann J.; Aubert, Xavier L.; Besling, Stefan; Dugast, Christian; Essen, Ute; Geller, Dieter; Haeb-Umbach, Reinhold; Kneser, Reinhard; Meier, Hans Günter; Oerder, Martin; Tran, Bach Hiep (Philips Journal of Research)
This paper gives an overview of the Philips Research system for continuous-speech recognition. The recognition architecture is based on an integrated statistical approach. The system has been successfully applied to various tasks in American English and German, ranging from small vocabulary tasks to very large vocabulary tasks and from recognition only to speech understanding. Here, we concentrate on phoneme-based continuous-speech recognition for large vocabulary recognition as used for dictation, which covers a significant part of our research work on speech recognition. We describe this task and report on experimental results. In order to allow a comparison with the performance of other systems, a section with an evaluation on the standard North American Business news (NAB2) task (dictation of American English newspaper text) is supplied.

Continuous speech dictation - From theory to practice - Steinbiss, Volker; Ney, Hermann J.; Essen, Ute; Tran, Bach Hiep; Aubert, Xavier L.; Dugast, Christian; Kneser, Reinhard; Meier, Hans Günter; Oerder, Martin; Haeb-Umbach, Reinhold; Geller, Dieter; Hoellerbauer, W.; et al. (Speech Communication)
This paper gives an overview of the Philips research system for phoneme-based, large-vocabulary, continuousspeech recognition. The system has been successfully applied to various tasks in the German and (American) English languages, ranging from small vocabulary tasks to very large vocabulary tasks. Here, we concentrate on continuousspeech recognition for dictation in real applications, the dictation of legal reports and radiology reports in German. We describe this task and report on experimental results. We also describe a commercial PC-based dictation system which includes a PC implementation of our scientific recognition prototype. In order to allow for a comparison with the performance of other systems, a section with an evaluation on the standard Wall Street Journal task (dictation of American English newspaper text) is supplied. The recognition architecture is based on an integrated statistical approach. We describe the characteristic features of the system as opposed to other systems: 1. the Viterbi criterion is consistently applied both in training and testing; 2. continuous mixture densities are used without tying or smoothing; 3. time-synchronous beam search in connection with a phoneme look-ahead is applied to a tree-organized lexicon.

Improvements in beam search for 10000-word continuous-speech recognition - Haeb-Umbach, Reinhold; Ney, Hermann (IEEE Transactions on Speech and Audio Processing)
The authors describe the improvements in a time-synchronous beam search strategy for a 10000-word continuous-speech recognition task. Basically they introduced two measures, namely a tree organization of the pronunciation lexicon and a novel look-ahead technique at the phoneme level. The experimental tests performed showed that the number of state hypotheses could be reduced from 50000 to 3000, i.e., by a factor of about 17. At the same time, the word error rate did not increase.

Progress in Large-Vocabulary, Continuous Speech Recognition - Ney, Hermann; Steinbeiss, Volker; Aubert, Xavier L.; Haeb-Umbach, Reinhold (Artifical Intelligence, Progress and Prospects of Speech Research and Technology, Munich)

An Overview of the Philips Research System for Large Vocabulary Continuous Speech Recognition - Ney, Hermann; Steinbeiss, Volker; Haeb-Umbach, Reinhold; Tran, Bach Hiep (International Journal on Pattern Recognition and Artificial Intelligence)
This paper gives an overview of a research system for phoneme based, large vocabulary continuous speech recognition. The system to be described has been applied to the SPICOS task, the DARPA RM task and a 12000 word dictation task. Experimental results for these three tasks will be presented. Like many other systems, the recognition architecture is based on an integrated statistical approach. In this paper, we describe the characteristic features of the system as opposed to other systems: (1) The Viterbi criterion is consistently applied both in training and testing. (2) Continuous mixture densities are used without any tying or smoothing; this approach can be viewed as a sort of statistical template matching. (3) Time-synchronous beam search is used consistently throughout all tasks; extensions using a tree organization of the vocabulary and phoneme lookahead are presented so that a 12000 word task can be handled.

Continuous Mixture Densities and Linear Discriminant Analysis for Improved Context-Dependent Acoustic Models - Aubert, Xavier L.; Haeb-Umbach, Reinhold; Ney, Hermann (ICASSP, Minneapolis)
Linear discriminant analysis (LDA) experiments reported previously (ICASSP-92 vol.1, p.13-16), are extended to context-dependent models and speaker-independent large vocabulary continuous speech recognition. Two variants of using mixture densities are compared: state-specific modeling and the monophone-tying approach where densities are shared across the states relevant to the same phoneme. Results are presented on the DARPA Resource Management (RM) task for both speaker-dependent (SD) and speaker-independent (SI) parts. Using triphone models based on LDA and continuous mixture densities, significant improvements have been observed and the following word error rates have been achieved: for the SD part, 7.8% without grammar and 1.5% with word pair; and for the SI part, 17.2% and 4.6%, respectively. These scores are averaged over 1200 SD or SI evaluation sentences and are among the best published so far on the RM database.

Design and use of speech recognition algorithms for a mobile radio telephone - Dobler, Stefan; Geller, Dieter; Haeb-Umbach, Reinhold; Meyer, Peter; Ney, Hermann; Ruehl, Hans Wilhelm (Speech Communication)
To decrease the hazards of using mobile phones while driving, voice processing provides several tools that simplify their use: echo cancellation allows comfortable hands-free conversation, feedback and user guidance by voice allow to operate the phone in eyes-busy situations, and last not least speech recognition frees from keypad data entry to operate the telephone. A comprehensive view of a device incorporating the above mentioned technologies, which has been realized as an add-on for the Philips car telephone family, will be presented. Emphasis is placed on the speech recognition algorithms. Robustness of the algorithms to changing acoustic environment was improved by estimating and subtracting the long-term spectrum. We will show that, if this operation is done recursively, it is equivalent to the high-pass filtering or RASTA (Relative Spectral Approaches) methods recently proposed in the literature.

Improvements in Connected Digit Recognition Using Linear Discriminant Analysis and Mixture Densities - Haeb-Umbach, Reinhold; Geller, Dieter; Ney, Hermann (ICASSP, Minneapolis)
Four methods were used to reduce the error rate of a continuous-density hidden Markov-model-based speech recognizer on the TI/NIST connected-digits recognition task. Energy thresholding sets a lower limit on the energy in each frequency channel to suppress spurious distortion accumulation caused by random noise. This led to an improvement in error rate by 15%. Spectrum normalization was used to compensate for across-speaker variations, resulting in an additional improvement by 20%. The acoustic resolution was increased up to 32 component densities per mixture. Each doubling of the number of component densities yielded a reduction in error rate by roughly 20%. Linear discriminant analysis was used for improved feature selection. A single class-independent transformation matrix was applied to a large input vector consisting of several adjacent frames, resulting in an improvement by 20% for high acoustic resolution. The final string error rate was 0.84%.

The Philips Research System for Large-Vocabulary Continuous-Speech Recognition - Steinbiss, Volker; Ney, Hermann; Haeb-Umbach, Reinhold; Train, B.-H.; Essen, U.; Kneser, Reinhard; Oerder, Martin; Meier, H. G.; Aubert, X.; Dugast, Christian; Geller, Dieter; Hoellerbauer, W.; et al. (EUROSPEECH, Berlin)
This paper gives a status report of the Philips research system for phoneme-based, large-vocabulary, continuous-speech recognition. Like for many other systems, the recognition architecture is based on an integrated statistical approach. We describe the characteristic features of the system as opposed to other systems: 1. The Viterbi criterion is consistently applied both in training and testing. 2. Continuous mixture densities are used without tying or smoothing. 3. Time-synchronous beam search in connection with a phoneme look-ahead is applied to a tree-organized lexicon. The system has been successfully applied to the American English DARPA RM task. Here, we report experimental results for a German 13 000-word Philips internal dictation task. In addition to the scientific prototype, a PC version has been set up which is described here for the first time.

The Philips Research System for Continuous-Speech Recognition: Overview and Evaluation on the DRAPA RM Task - Aubert, Xavier L.; Ney, Hermann; Haeb-Umbach, Reinhold (DRAPA Speech Workshop)

Improvements in Speech Recognition for Voice Dialling in the Car Environment - Geller, Dieter; Haeb-Umbach, Reinhold; Ney, Hermann (ESCA Workshop on Speech Recognition in Adverse Conditions, Cannes-Mandelieu)

A modified trellis coding technique for partial response channels - Haeb-Umbach, Reinhold (IEEE Transactions on Communications)
The problem of trellis coding for multilevel baseband transmission over partial response channels with transfer polynomials of the form (1+or-D/sup N/) is addressed. The novel method presented here accounts for the channel memory by using multidimensional signal sets and partitioning the signal set present at the noiseless channel output. It is shown that this coding technique can be viewed as a generalization of a well-known procedure for binary signaling: the concatenation of convolutional codes and inner block codes that are tuned to the channel polynomial. It results in high coding gains with moderate complexity if some bandwidth expansion is accepted.

Trellis codes for partial-response magnetooptical direct overwrite recording - Haeb-Umbach, Reinhold; Lynch, Robert (IEEE Journal on Selected Areas in Communications)
The authors present conditions on the error sequences between channel input sequences which guarantee certain lower bounds on the free Euclidian distance at the output of a partial-response (PR) class I or II channel. From these expressions, trellis codes are derived which improve performance of binary signaling over noisy PR channels with reduced complexity maximum-likelihood sequence detection. They are shown to be compatible with the input restriction caused by the magnetooptical resonant coil direct overwrite recording scheme. The codes achieve high signal-to-noise ratio coding gains of 3 dB (on PR class I) and 2.2 dB (on PR class II) with rates as close to, but strictly less than, the capacity of the initial input restriction as desired. The performance of these codes is analyzed with an optical channel simulation system which shows that one code has the rare but highly desirable property that its maximum-likelihood sequence detector (MLSD) is less complex than the MLSD of the reference system and still achieves an error rate performance gain of 1.8 dB

Linear Discriminant Analysis for Improved Large Vocabulary Continuous Speech Recognition - Haeb-Umbach, Reinhold; Ney, Hermann (ICASSP, San Francisco)
The interaction of linear discriminant analysis (LDA) and a modeling approach using continuous Laplacian mixture density HMM is studied experimentally. The largest improvements in speech recognition could be obtained when the classes for the LDA transform were defined to be sub-phone units. On a 12000 word German recognition task with small overlap between training and test vocabulary a reduction in error rate by one-fifth was achieved compared to the case without LDA. On the development set of the DARPA RM1 task the error rate was reduced by one-third. For the DARPA speaker-dependent no-grammar case, the error rate averaged over 12 speakers was 9.9%. This was achieved with a recognizer using LDA and a set of only 47 Viterbi-trained context-independent phonemes.

Improvements in Beam Search for 10,000-Word-Continuous Speech Recognition - Ney, Hermann; Haeb-Umbach, Reinhold; Tran, B. H.; Oerder, Martin (ICASSP, San Francisco)
This paper describes the improvements in a time synchronous beam search strategy for a 10000-word continuous speech recognition task. The improvements are based on two measures: a tree-organization of the pronunciation lexicon and a novel look-ahead technique at the phoneme level, both of which interact directly with the detailed search at the state levels of the phoneme models. Experimental tests were performed for four speakers on a 12306-word task. As a result of the above measures, the overall search effort was reduced by a factor of 17 without a loss in recognition accuracy.

A Look-Ahead Technique for Large Vocabulary Continuous Speech Recognition - Haeb-Umbach, Reinhold; Ney, Hermann (EUROSPEECH, Genova)
In a large vocabulary continuous speech recognition task the search for the "best" (in the maximum-a-posteriori sense) word sequence is the most (computing) time consuming part of the system. End-of-word hypotheses are created almost every time frame. With a stochastic language model every lexicon entry is an admissible successor candidate. By using a "fast match" module which scores the word candidates according to their acoustic feasibility ahead of the current time frame, the search cost can be considerably reduced. Only the fraction of the words with favourable fast match scores will be further processed in the detailed match, where the likelihood of a segment of acoustics given the word model is computed. We derive a novel word selection strategy which is "consistent" in the sense that it introduces no additional decoding errors and which still reduces the search space by a factor of 2 - 3 compared to standard Viterbi beam search. Giving up the consistency requirement, pruning strategies can be deduced which further reduce the search effort significantly: the size of the word startup list is reduced to 2% - 4% of its original size with a modest increase in error rate by l%-2%.

Coding and Signal Processing for a Magneto-optic Resonant Bias Coil Overwrite Experiment - Haeb-Umbach, Reinhold; Rugar, D.; Howell, T.; Coleman, G. P. (International Conference on Communication, Atlanta)
The resonant bias coil technique requires even-numbered transitions to be written on even-numbered clock cycles, and odd-numbered transitions on odd-numbered clock cycles. This constraint is met by double-spaced run-length-limited (d, k, 2) codes where the number of consecutive zeros is even. Exploiting the polarity of the transitions, a detection window that has double the size of the code bit period is possible. The authors describe a detection circuit that achieves the enlarged detection window, and present a phase detector which is particularly simple to implement. Channel bit-error-rate measurements have been carried out employing a rate 1/3 (2, 8, 2) code. Error rates of 10/sup -8/ were achieved for recording densities up to 33 kb/in. The results demonstrate the expected excellent immunity of the direct overwrite scheme to bloom.

A systematic approach to carrier recovery and detection of digitally phase modulated signals on fading channels - Haeb-Umbach, Reinhold; Meyr, Heinrich (IEEE Transactions on Communications)
The problem of optimal carrier recovery and detection of digitally phase modulated signals on fading channels by using a nonstructured approach is presented, i.e. no constraint is placed on the receiver structure. First, the optimal receiver is derived for digitally phase-modulated signals when transmitted over a frequency-nonselective fading channel with memory. The memory results from the fact that usually the coherence time of the channel is larger than the symbol period. Symbols adjacent in time cannot be detected independently and therefore the well-known quadratic receiver is not optimal in this case. A maximum a posteriori (MAP) detector is derived and explicitly utilizes the channel memory for carrier recovery. The derivation shows that the optimal carrier recovery is, under certain conditions, a Kalman filter. Some attractive properties of this carrier recovery unit (including the absence of hang up) are discussed. Then the error rate of several digital modulation schemes is calculated taking the performance of the filter into account. The differences in susceptibility of the modulation schemes to carrier phase jitter are specified.

A Comparison of Coherent and Differentially Coherent Detection Schemes for Fading Channels - Haeb-Umbach, Reinhold (International Conference on Vehicular Technology, Philadelphia)
The common basis of coherent and differentially coherent detection is considered from the viewpoint of carrier recovery as the estimation of fading distortion. In the differentially coherent receiver a simple estimate of the fading distortion is used whereas the coherent receiver uses the optimal estimate. The bit error rates (BERs) for M-ary PSK (phase-shift keyed) and DPSK (double phase-shift keyed) transmission are calculated using a single method of calculation for both detection schemes. The calculation takes into account nonperfect carrier recovery, cochannel interference, and diversity. The results allow a direct comparison of the two schemes and show that coherent detection is preferable in many realistic fading environments.

A digital Synchronizer for Linearly Modulated Signals Transmitted over a Frequency-Nonselective Fading Channel - Haeb-Umbach, Reinhold; Meyr, Heinrich (International Conference on Communications, Philadelphia)
A digital carrier recovery structure which allows coherent detection on frequency-nonselective fading channels is presented. The synchronizer estimates the multiplicative distortion introduced by the channel. It is shown that the structure is superior to a phase-locked loop and is well suited for a fully digital realization. A detailed synchronizer design and simulation results are presented for a land-mobile radio channel. This includes a novel scheme for a fully digital frequency offset estimation and correction.

Optimal Carrier Recovery and Detection on Frequency-Nonselective Fading Channels - Haeb-Umbach, Reinhold; Meyr, Heinrich (Proc. Symposium on Inf. Theory and Appl. (SITA), Tokio)

An all digital implementation of a receiver for bandwidth-efficient communication - Oerder, Martin; Ascheid, Gerd; Haeb-Umbach, Reinhold; Meyr, Heinrich (Signal Processing: Theories and Applications)