A comparison of time-domain time-scale modification algorithms

Time-domain approaches to time-scale modification are popular due to their ability to produce high quality results at a relatively low computational cost. Within the category of time-domain implementations quite a number of alternatives exist, each with their own computational requirements and associated output quality. This paper provides a computational and objective output quality assessment of a number of popular time-domain time-scaling implementations; thus providing a means for developers to identify a suitable algorithm for their application of interest. In addition, the issues that should be considered in developing time-domain algorithms are outlined, purely in the context of a waveform editing procedure

A comparison of time-domain time-scale modification algorithms

Time-domain approaches to time-scale modification are popular due to their ability to produce high quality results at a relatively low computational cost. Within the category of time-domain implementations quite a number of alternatives exist, each with their own computational requirements and associated output quality. This paper provides a computational and objective output quality assessment of a number of popular time-domain time-scaling implementations; thus providing a means for developers to identify a suitable algorithm for their application of interest. In addition, the issues that should be considered in developing time-domain algorithms are outlined, purely in the context of a waveform editing procedure

Audio- ja puhesignaalien aika-asteikon muuttaminen

In audio time-scale modification (TSM), the duration of an audio recording is changed while retaining its local frequency content. In this thesis, a novel phase vocoder based technique for TSM was developed, which is based on the new concept of fuzzy classification of points in the time-frequency representation of an input signal. The points in the time-frequency representation are classified into three signal classes: tonalness, noisiness, and transientness. The information from the classification is used to preserve the distinct nature of these components during modification. The quality of the proposed method was evaluated by means of a listening test. The proposed method scored slightly higher than a state-of-the-art academic TSM technique, and similarly as a commercial TSM software. The proposed method is suitable for high-quality TSM of a wide variety of audio and speech signals.Äänen aika-asteikon muuttamisessa äänitteen pituutta muokataan niin, että sen paikallinen taajuussisältö säilyy samanlaisena. Tässä diplomityössä kehitettiin uusi, vaihevokooderiin pohjautuva menetelmä äänen aika-asteikon muuttamiseen. Menetelmä perustuu äänen aikataajuusesityksen pisteiden sumeaan luokitteluun. Pisteet luokitellaan soinnillisiksi, kohinaisiksi ja transienttisiksi määrittämällä jatkuva totuusarvo pisteen kuulumiselle kuhunkin näistä luokista. Sumeasta luokittelusta saatua tietoa käytetään hyväksi näiden erilaisten signaalikomponenttien ominaisuuksien säilyttämiseen aika-asteikon muuttamisessa. Esitellyn menetelmän laatua arvioitiin kuuntelukokeen avulla. Esitelty menetelmä sai kokeessa hieman paremmat pisteet kuin viimeisintä tekniikkaa edustava akateeminen menetelmä, ja samanlaiset pisteet kuin kaupallinen ohjelmisto. Esitelty menetelmä soveltuu monenlaisien musiikki- ja puhesignaalien aika-asteikon muuttamiseen

Interactively skimming recorded speech

Thesis (Ph. D.)--Massachusetts Institute of Technology, Program in Media Arts & Sciences, 1994.Includes bibliographical references (p. 143-156).Barry Michael Arons.Ph.D

DEVELOPMENT AND EVALUATION OF ENVELOPE, SPECTRAL AND TIME ENHANCEMENT ALGORITHMS FOR AUDITORY NEUROPATHY

Auditory neuropathy (AN) is a hearing disorder that reduces the ability to detect temporal cues in speech, thus leading to deprived speech perception. Traditional amplification and frequency shifting techniques used in modern hearing aids are not suitable to assist individuals with AN due to the unique symptoms that result from the disorder. This study proposes a method for combining both speech envelope enhancement and time scaling to combine the proven benefits of each algorithm. In addition, spectral enhancement is cascaded with envelope and time enhancement to address the poor frequency discrimination in AN. The proposed speech enhancement strategy was evaluated using an AN simulator with normal hearing listeners under varying degrees of AN severity. The results showed a significant increase in word recognition scores for time scaling and envelope enhancement over envelope enhancement alone. Furthermore, the addition of spectral enhancement resulted in further increase in word recognition at profound AN severity

Efficiency in audio processing : filter banks and transcoding

Audio transcoding is the conversion of digital audio from one compressed form A to another compressed form B, where A and B have different compression properties, such as a different bit-rate, sampling frequency or compression method. This is typically achieved by decoding A to an intermediate uncompressed form, and then encoding it to B. A significant portion of the involved computational effort pertains to operating the synthesis filter bank, which is an important processing block in the decoding stage, and the analysis filter bank, which is an important processing block in the encoding stage. This thesis presents methods for efficient implementations of filter banks and audio transcoders, and is separated into two main parts. In the first part, a new class of Frequency Response Masking (FRM) filter banks is introduced. These filter banks are usually characterized by comprising a tree-structured cascade of subfilters, which have small individual filter lengths. Methods of complexity reduction are proposed for the scenarios when the filter banks are operated in single-rate mode, and when they are operated in multirate mode; and for the scenarios when the input signal is real-valued, and when it is complex-valued. An efficient variable bandwidth FRM filter bank is designed by using signed-powers-of-two reduction of its subfilter coefficients. Our design has a complexity an order lower than that of an octave filter bank with the same specifications. In the second part, the audio transcoding process is analyzed. Audio transcoding is modeled as a cascaded quantization process, and the cascaded quantization of an input signal is analyzed under different conditions, for the MPEG 1 Layer 2 and MP3 compression methods. One condition is the input-to-output delay of the transcoder, which is known to have an impact on the audio quality of the transcoded material. Methods to reduce the error in a cascaded quantization process are also proposed. An ultra-fast MP3 transcoder that requires only integer operations is proposed and implemented in software. Our implementation shows an improvement by a factor of 5 to 16 over other best known transcoders in terms of execution speed

A transient-preserving audio time-stretching algorithm and a real-time realization for a commercial music product

The core of this work is a sub-band transient detection/preservation scheme based on the complex domain transient detection, and inspired by Robel’s work. This proposed technique can be integrated in a real-time phase vocoder analysis/synthesis scheme without introducing latency at relatively low computational cost

Engineering systematic musicology : methods and services for computational and empirical music research

One of the main research questions of *systematic musicology* is concerned with how people make sense of their musical environment. It is concerned with signification and meaning-formation and relates musical structures to effects of music. These fundamental aspects can be approached from many different directions. One could take a cultural perspective where music is considered a phenomenon of human expression, firmly embedded in tradition. Another approach would be a cognitive perspective, where music is considered as an acoustical signal of which perception involves categorizations linked to representations and learning. A performance perspective where music is the outcome of human interaction is also an equally valid view. To understand a phenomenon combining multiple perspectives often makes sense. The methods employed within each of these approaches turn questions into concrete musicological research projects. It is safe to say that today many of these methods draw upon digital data and tools. Some of those general methods are feature extraction from audio and movement signals, machine learning, classification and statistics. However, the problem is that, very often, the *empirical and computational methods require technical solutions* beyond the skills of researchers that typically have a humanities background. At that point, these researchers need access to specialized technical knowledge to advance their research. My PhD-work should be seen within the context of that tradition. In many respects I adopt a problem-solving attitude to problems that are posed by research in systematic musicology. This work *explores solutions that are relevant for systematic musicology*. It does this by engineering solutions for measurement problems in empirical research and developing research software which facilitates computational research. These solutions are placed in an engineering-humanities plane. The first axis of the plane contrasts *services* with *methods*. Methods *in* systematic musicology propose ways to generate new insights in music related phenomena or contribute to how research can be done. Services *for* systematic musicology, on the other hand, support or automate research tasks which allow to change the scope of research. A shift in scope allows researchers to cope with larger data sets which offers a broader view on the phenomenon. The second axis indicates how important Music Information Retrieval (MIR) techniques are in a solution. MIR-techniques are contrasted with various techniques to support empirical research. My research resulted in a total of thirteen solutions which are placed in this plane. The description of seven of these are bundled in this dissertation. Three fall into the methods category and four in the services category. For example Tarsos presents a method to compare performance practice with theoretical scales on a large scale. SyncSink is an example of a service

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies

A survey on hardware and software solutions for multimodal wearable assistive devices targeting the visually impaired

The market penetration of user-centric assistive devices has rapidly increased in the past decades. Growth in computational power, accessibility, and cognitive device capabilities have been accompanied by significant reductions in weight, size, and price, as a result of which mobile and wearable equipment are becoming part of our everyday life. In this context, a key focus of development has been on rehabilitation engineering and on developing assistive technologies targeting people with various disabilities, including hearing loss, visual impairments and others. Applications range from simple health monitoring such as sport activity trackers, through medical applications including sensory (e.g. hearing) aids and real-time monitoring of life functions, to task-oriented tools such as navigational devices for the blind. This paper provides an overview of recent trends in software and hardware-based signal processing relevant to the development of wearable assistive solutions