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Present day secure speech communication mainly deals with providing maximum security at the cost of minimum complexity. While doing so, they are mainly looking forward to the speech coders for compressing the speech signal thus minimizing the complexity. In this paper an algorithm is proposed that aims at exploiting the basic characteristics of speech signal, while designing such system with reduced...
Voiced speech is produced by excitation of the vocal tract system with the quasiperiodic vibrations of the vocal folds at the glottis. These excitations have become significantly stronger when the vocal folds are fully opened or about to be closed. In this work, the focus is on estimating these instants of significant excitation using temporal phase periodicity present in the speech signal. Assuming...
This paper presents a novel scheme of watermarking of digital images for copyright protection and authentication. In this paper we proposed a method of embedding owner's speech signal. Speech being a biometric data, the watermark signal in this method is expected to be more meaningful and has closer correlation with copyright holder. The main issue of concern here is the capacity because the speech...
The amount of speaker specific information in speech signal varies from frame to frame depending on spoken text and environmental conditions. A frame selection at the preprocessing stage can be an added advantage in this context. In pre-quantization (PQ) we select a new sequence of frames Y from the original frames X such that length of Y is less than X. In this paper, we first analyze a number of...
Inadequate velopharyngeal closure, due to structural or neurological problems, allows air to pass through the nasal cavity leading to introduction of inappropriate nasal resonances during speech production resulting in hypernasal speech. Our previous work on the acoustic analysis of hypernasal speech using group delay function for the detection of hypernasality showed stable effects of vowel nasalization...
The pioneering work on the `separation of speech from mixture of acoustic sources' dates back to as early as 70s and since then, two main approaches namely traditional approach using signal-processing techniques and computational auditory scene analysis (CASA) approach using auditory-modeling methods have been concurrently attempted by researchers to find solution to the problem of what is known as...
The performance of a speaker recognition system decreases when the speaker is under stress or emotion. In this paper we explore and identify a mechanism that enables use of inherent stress-in-speech or speaking style information present in speech of a person as additional cues for speaker recognition. We quantify the the inherent stress present in the speech of a speaker mainly using 3 features, namely,...
A common approach in mapping a signal to discrete events is to define a set of symbols that correspond to useful acoustic features of the signal over a short constant time interval. This paper proposes a hidden Markov models (HMM) based speech recognition by using cepstrum feature of the signal over adaptive time interval. First pitch period is detected by dyadic wavelet transform and divides the...
In classification tasks, the error rate is proportional to the commonality among classes. Conventional GMM-based modeling techniques fail to capture the unique features of a class. Classification accuracy can be improved if the modeling technique is able to capture the unique features of each class. For any two models and their corresponding training data, the log-likelihoods may be assumed to be...
A number of techniques have been proposed in the literature for phoneme based speech recognition system. In this paper, a technique for automatic phoneme recognition using zero-crossings (ZC) and magnitude sum function (MSF) is proposed. The number of zero-crossings and magnitude sum function per frame are extracted and a minimum distance classifier is proposed to recognize the phonemes in each frame...
Sound localization systems (SLS) identify the direction of a sound source. However, most of approaches focus on near-field identification, i.e. 1~2 m. In this paper we develop a novel algorithm for far-field sound localization based on the average magnitude difference function (AMDF), thereby extending the distance to 5 m. The far-field SLS is implemented on a field programmable gate array (FPGA)...
Cochlear implants (CI) are electrical prosthesis that partially replaces the function of the human ear. As cochlear implants are system specific there is a need for simulation of the system parameters prior to implantation. In the current work we have considered the system specific parameters like number of channels, type of filters by developing uniform bandwidth filter based acoustic CI model and...
Multi pattern Viterbi algorithm (MPVA) to jointly decode and recognize multiple speech patterns for automatic speech recognition (ASR) is proposed. The MPVA is a generalization of the Viterbi algorithm (VA) to jointly decode multiple patterns for a given standard hidden Markov model (HMM). Unlike our previously proposed constrained multi pattern Viterbi algorithm (CMPVA), the MPVA does not require...
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