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This paper focuses on the quality of speech coding parameters extraction under noisy and clean conditions. The influence of speech enhancement on the quality of extracted parameters for a low bit rate speech coder is addressed. MELP vocoder is used to estimate three parameters: the fundamental frequency, voicing and linear prediction coefficients. De-noising methods in MELPe vocoder and SMV are adopted...
The influence of noise on the performance of a low bit rate parametric speech coder is addressed in this paper. MELP vocoder is used to estimate three parameters: the fundamental frequency, voicing and linear prediction coefficients. Influence of different noises under various acoustic environments on the MELP vocoder's parameters is studied. Pitch accuracy rate, voicing decision error rate and average...
With the development of network and speech coding, various applications of transmitting voice over internet protocol (VoIP) come into our lives. Speech reconstruction is required, as the packet loss caused by the real-time constraints transmission protocols results in the loss of speech segments. The paper proposes a novel approach to addressing speech reconstruction problem utilizing a high-order...
In mobile communications, environmental noise often reduces the quality and intelligibility of speech. Problems caused by far-end noise, in the sending side of the communication channel, can be alleviated by using a noise reducing preprocessing stage before the encoder. In this study, a modification increasing the robustness of the encoder itself to background noise is proposed. Specifically, by using...
At low bit-rates, speech coders relying on a single source model have difficulties to properly render speech and background noise simultaneously. Difficulties can get even bigger when using speech enhancement techniques within the coding scheme: these have shown to improve quality for clean speech, but introduce unpleasant instabilities under noisy conditions. This paper presents a novel approach...
In this paper, we propose a speech enhancement approach integrating codebook and Wiener filtering. Codebook-based method could reserve speech spectrum well, but leaves a large amount of residual noise. Signal to noise ratio (SNR)-estimation-based Wiener filtering method could reduce noise well, but leads to relatively serious speech distortion. To solve the problem, the gain functions of the above...
The majority of speech coding algorithms are based on the code excited linear prediction (CELP) paradigm, modelling the speech signal by linear prediction. This coding approach offers the advantage of a very short algorithmic delay, due to the windowing scheme based on rectangular windowing of the residual of the linear predictor. Although widely used, the performance and structural choices of this...
With wireless acoustic sensor network extending to the services like surveillance of sensitive areas, such as Line of Control, or unmanned terrains, interest in robust, narrowband and low bit rate speech codecs is increasing. This has resulted in a need for evaluation of such codecs. This work investigates the different factors like bit rate, algorithmic delay, implementation, and more importantly...
This paper investigates the use of a signal enhancement technique, namely a noise suppressing nonlinearity, on the adaptive filter error in order to increase the stability and the performance of acoustic echo cancellation (AEC) when there is a continuous distortion to the acoustic echo signal. The algorithm presented here differs from others in that the enhancement of signal is done in the adaptation...
CELP schemes with trained excitation codebook are able to reproduce more complex waveforms than stochastic CELP schemes. Here we present a new algorithm for the design of trained CELP excitation code-books which are well adapted to the residual of speech even in transition regions. The vectors of the excitation codebook are adapted to a training speech sequence by applying an iterative algorithm....
Voiced speech is characterized by a high level of periodicity. In order to encode voiced speech with a good quality, the correct degree of periodicity must be preserved. The proposed coding algorithm attempts to reproduce the correct level of periodicity even at low bit rates. The method exploits the temporal redundancy of voiced segments in order to achieve high compression rates. Voiced speech is...
This paper presents the performance evaluation of different speech coding techniques in wireless packet switching networks: the goal of our study is to increase network capacity while maintaining a smooth degradation of quality at high loads and heavy interference, in order to make it possible for different kinds of information to coexist in a single network infrastructure. In the paper we propose...
Several state-of-the-art switched audio codecs employ the closed-loop mode decision to select the best coding mode at every frame. The closed-loop mode selection is known to have good performance but also high complexity. The new approach we propose in this paper is a low-complexity version of the closed-loop approach, based on similar decisions which compute the coding distortion of each mode and...
A novel noise-robust soft Voice Activity Detector (VAD) operating in the short-time Fourier domain is presented. A speech energy gain is obtained by frame-wise processing of a noisy speech signal with a speech codebook algorithm. This gain can be used for robust voice detection. A speaker-independent speech codebook, consisting of spectral envelopes, is created in the training process. While applying...
A practical approach for the design of multiple-description scalar quantization of speech is presented that conforms to standard G.711 PCM. The method chiefly consists of an index assignment algorithm that enables the side decoders to exhibit SNR characteristics comparable to those of the standard logarithmic quantizer. With two-channel transmission of multiple descriptions, an increase in robustness...
This paper discusses the design and implementation of a low bit rate codec along with its performance at different bit rates. The International Telecommunications Union's G.728 CELP speech coder is specifically designed for low coding delay and toll quality speech at a rate of 16kbps. Here, we present the design of a CELP(Code Excited Linear Prediction) algorithm similar to the above coder, but can...
Medium bit rate hybrid speech coding schemes have gained much interest in the recent years and many of them have been standardized for various applications. This work characterizes a speech codec in a Compressive Sensing (CS) framework. We mainly demonstrate two aspects 1) Simultaneous compression and de-noising of speech by CS 2) Appropriate quantization of CS measurements to design medium bit-rate...
Methods applied to ensure privacy of digital data became essential in many applications. In this paper, we propose a Fast Fourier transform based steganography technique that utilizes wideband speech as a cover to hide image. Selected locations from high-frequencies of the magnitude speech spectrum are exploited to embed the digital image. Naturalness and intelligibility of the resulted (stego) speech...
In this paper, we propose a MDCT domain postfilter to suppress the noise in speech spectral valleys and allow more noise in speech spectral peaks. Since peaks and valleys of MDCT coefficients don't correspond to those of speech spectrum, we construct both the short-term and long-term postfilter using MDCT pseudo spectrum. The short-term postfilter is induced by smoothing the envelope that is obtained...
Speech intelligibility for noise reduction algorithms which were integrated into perceptual wavelet packet-based speech coding strategy in cochlear implant (CI) processors was investigated in this study. The noise reduction algorithms including time-adaptive wavelet thresholding (TAWT) and time-frequency spectral subtraction (TFSS) were selected for this study due to simple and suitable for real-time...
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