Automatic speech recognition from multiple distant microphones poses significant challenges because of noise and reverberations. The quality of speech acquisition may vary between microphones because of movements of speakers and channel distortions. This paper proposes a channel selection approach for selecting reliable channels based on selection criterion operating in the short-term modulatio...

In this paper, we propose a two-stage speech enhancement technique. In the training stage, a Gaussian Mixture Model (GMM) of the mel-frequency cepstral coefficients (MFCCs) of a user’s clean speech is computed wherein the component densities of the GMM serve to model the user’s “acoustic classes.” In the enhancement stage, MFCCs from a noisy speech signal are computed and the underlying clean a...

This paper proposes an exemplar-based speech enhancement method based on high-resolution STFT magnitude spectrograms, where a selection of the nonnegative training data is used as the dictionary to provide a holistic nonnegative representation of the test data. We discuss how this exemplar-based model ensures that the enhanced speech signal falls on the speech manifold, which improves the quali...

This paper presents a forward error correction (FEC) technique based on a multipulse representation of the excitation for codeexcited linear prediction (CELP) speech transmission under packet loss conditions. In this approach, the encoder sends the position of a pulse that it is used for the resynchronization of the adaptive codebook, so that propagation errors can be prevented. At the decoder,...

This paper proposes an objective quality assessment method for voice communication systems using packet loss information. Based on the fact that the effect of packet loss to perceptual quality varies depending on the signal characteristics of lost packets, different weighting factors are applied to predicting overall quality. Considering the key paradigm of low bit rate speech coders such as pa...

In mobile devices, perceived speech signal degrades significantly in the presence of background noise as it reaches directly at the listener's ears. There is a need to improve the intelligibility and quality of the received speech signal in noisy environments by incorporating speech enhancement algorithms. This paper focuses on speech enhancement method including auditory masking propertie...

Vector Quantization (VQ) has been extensively used in speech vocoders. Phase information is often ignored or coarsely represented in parametric coders because of the difficulties facing phase quantization. This paper introduces a novel distortion measure for the low-band speech signal that takes phase information into consideration, with no increase in the bit-rate. This measure has been used i...

The voice signal in Voice over Internet protocol (VoIP) system is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss jitter. The work in this paper presents the implementation of finite impulse response (FIR) filter for voice quality improvement in the VoIP system through distributed arithmetic (DA) algorithm. The VoIP...

In this paper, we present a new adaptive microphone array processing algorithm for hands-free sound capture. Most traditional adaptive beamforming techniques operate solely on the basis of the direction of arrival of the desired source and are blind to any knowledge about the signal itself. In contrast, the proposed algorithm uses a nonlinear spatial filter to generate an estimate of the magnit...

The quality of speech signal over a VoIP system is degraded by various network layer problems which include jamming, jitter, packet loss. Different types of noises also degrade the quality of speech signal such as external noise and quantization noise. This paper improves the quality of VoIP speech signal affected by these noises and network layer problems. The quality of degraded VoIP speech s...