In human-computer interaction, humans can interact with the computer help of text, audio, images, speech, etc. Interacting using speech recognitions in particularly audio segmentation is a challenging task due to accent or way pronouncing style. To input mathematical symbols, words, functions, and expressions keyboard are tedious time-consuming. Input this speeds up process. paper, an SBTI (aud...

We present a framework for the analysis and synthesis of acoustical instruments based on data driven probabilistic inference modeling Audio time series and boundary conditions of a played instrument are recorded and the non linear mapping from the control data into the audio space is inferred using the general inference framework of Cluster Weighted Modeling The resulting model is used for real...

This paper describes the design of an IEC986 audio decoder, and its CMOS implementation. The chip is designed to decode serial audio data in the IEC-986 audio format. Our chip decodes the input stream and produces two 21-bit parallel digital output signals, which can be connected to digital-to-analog converters. The design also includes a 5-bit variable gain feature, which can be used to select...

This longitudinal study investigated the impact of different Frequency Patterns (FP) on the syntactic production of a six-year-old EFL learner in a home context. Target syntactic constructions were presented using games and plays and were traced for their occurrence patterns in input and output. Following each instruction period, the constructions were measured through immediate and delayed ora...

The study of impulse response technology allows audio engineers to attempt to recreate the tonal characteristics of a certain device. The purpose of this project was to recreate a sonic representation of analog microphone pre amplifiers and acoustic spaces through the use of impulse response technology. By creating these impulses, the student will attain the sonic characteristics of professiona...

Speaker diarization is the task of partitioning an input stream into speaker homogeneous regions, or in other words, to determine ”who spoke when.” While approaches to this problem have traditionally relied entirely on the audio stream, the availability of accompanying video streams in recent diarization corpora has prompted the study of methods based on multimodal audio-visual features. In thi...

This paper describes the design and development of new graphical software for spectral audio processing. There are three phases to accomplish the audio processing. First, spectrograms are generated from an input audio signal using Short-Time Fourier Transform (STFT) analysis, and the sonogram image is continually shifting from right to left. Secondly, the sonogram is transformed by placing prep...

Marsyas is an open source software framework for audio analysis, synthesis and retrieval with specific emphasis on Music Information Retrieval. It is developed by an international team of programmers and researchers led by George Tzanetakis. In MIREX 2012 the Marsyas team participated in the following tasks that we have participated in the past: Audio Classical Composer Identification, Audio Ge...

The paper deals with design and performance analysis of algorithms that utilize parallel signal-processing methods and SIMD technology for multiply-and-add algorithm for digital audio signal processing. This algorithm is used for summing the gained input signals on output buses in applications for distributing, mixing, effect-processing, and switching multi-format digital audio signal in an aud...