Roberto Montagna

Some experiments of 7 kHz audio coding at 16 kbit/s

The problem of coding speech at 7 kHz is considered. A possible application of these codecs could be the videophone, especially for hands-free telephone use. The authors propose a split-band coder structure, where both hands are coded with analysis-by-synthesis techniques in order to take advantage of their high coding gain. Two techniques are considered: multipulse coding and codebook-excited linear prediction. The structures of two possible codecs are described, and indications are given of the considerations that led to their design. Their main characteristic is the use, as far as possible, of excitation model parameters optimized within the analysis by synthesis loop, still maintaining a reasonable computational complexity. Subjective and objective results, obtained with high-quality speech, are reported. They show that the obtained speech quality is close to the original.<<ETX>>

Method of and device for coding speech signals with analysis-by-synthesis techniques

The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, while the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

Comparison of two speech codecs for DMR systems

Abstract Several activities have been undertaken in Italy in the field of Digital Mobile Radio (DMR). In particular, concerning speech coding two codecs were developed to be compared on experirental Single Carrier Per Channel (SCPC) systems. These two codecs are described in the paper and their performances are compared. Both coders are based on a Split-Band (SB) structure that allows an easy exploitation of masking effects: they are a Sub-Band Coder (SBC) and SB-APC (Adaptive Predictive Coder). The main feature of both schemes is the use of Vector Quantization (VQ) to compact the side information. In addition the SB-APC scheme uses the technique of post-filtering to shape the quantizing noise of each sub-band. The main result reported in the paper is that the performances of the two schemes are almost equivalent although their structure is very different.

Selection phase of gsm half-rate channel

Aim of the paper is the description of the procedure adopted by ETSI to obtain the standard for the half rate channel of the GSM system. In particular, the paper describes the conditions to be tested to assess the performance of schemes proposed as candidates to the standard for the GSM half-rate speech channel. Selection rules were defined to rank the candidates taking into account the average speech quality, delay and complexity. Speech qualiby was measured by means of subjective tests that were performed according a suitable test plan designed with the aim of comparing the performance of the candidates with full-rate ones. Subjective testing of hardware prototypes needs a Host Laboratory Session to process the speech material through the candidates while emulating test conditions. The paper illustrates the basic structure of the Host Laboratory System and some information on its performance. Next, the main features of the submitted candidates are described and for each of them the results obtained in the subjective experiments. Subjective test data were collected and anlalysed: the results of the analysis are illustrated and main conclusions on the fullfillment of requirements reported.

Speech and image coding for digital communications

The paper is presenting a sketch of the main results achieved in the recent years and of the ongoing activities (mainly within the Standardization Bodies) in the areas of audio and video coding for bandwidth reduction. The evolution of the telecommunication Field and the needs which originate the different standards are also considered. The paper is organized in two main chapters: the first one is dealing with the techniques for audio and speech coding while the second presents the main achievements in terms of Recommendations and consolidate algorithms for image coding. Concerning speech and audio coding the paper presents and overview of the coding techniques and then examines the different environments of transmissions (network and mobile communication), the bandwidths required to satisfy the considered applications and the defined speech quality evaluation methods. A similar approach has been used for video coding and the evolution of the coding techniques appearing in the CCITT Recommendations is presented. The ongoing activities within CCIR, CMIT, CCITT and ISO towards new worldwide standards are also outlined providing indications on the areas which will require further developments.

A Two-Band CELP Audio Coder at 16 kbit/s and Its Evaluation

Future applications like HDTV video, multimedia, mobile audio-visual, N-ISDN and B-ISDN services will require wideband speech coding (7 KHz bandwidth). The leading idea, supported by some preliminary experiments on human perception, is that customers will prefer systems providing images with highly natural speech and also telephone services with wideband speech.

Vector quantization and perceptual criteria in SVD based CELP coders

Experiments in which singular value decomposition (SVD) is applied to code excited linear prediction (CELP) coders are presented. This means that the concept of selection of the optimum innovation pattern through an analysis by synthesis (ABS) method can be replaced by a weighted mean-squared-error computation in a transformed domain. The transformed signal is obtained by representing the residual signal on the orthonormal basis obtained by SVD of the matrix of the truncated linear predictive coding (LPC) filters impulse response. The weighting consists of the vector gain and the singular values (SVs) of the matrix. The characteristic of the SVD approach is that the excitation signal is represented as a linear combination of orthonormal signals whose spectra show characteristics quite similar to those of bandpass filters. Moreover, errors in the amplitude of each component at the filter input reflect errors only in the corresponding components at the filter output, weighted by the associated SV. This characteristic can be exploited by incorporating a simplified auditory model to determine the subjective importance of the singular value components.<<ETX>>

TCH-HS activities for the GSM channel standardization

The aim of the paper is the description of the procedure adopted by ETSI to obtain the standard for the half-rate channel of the GSM system; in particular, the paper describes: i) the conditions to be tested to assess the performance of schemes proposed as candidates, for the standard, and ii) selection rules defined to rank the candidates, taking into account: average speech quality, delay and complexity. Speech quality was measured by means of subjective tests performed according to suitable test plans designed with the aim of comparing the candidates with the GSM full-rate codec. Subjective testing of hardware prototypes needs a Host Laboratory Session to process the speech material through the candidates while emulating test conditions. The paper reports the basic structure of the Host Laboratory System (HLS) and some information on its performance. Next, some information on the types of analysis performed and on the results of the analysis are illustrated and main conclusions on the fulfilment of requirements are reported. At last, the paper deals with the DTX of the GSM half-rate channel. The paper is solicited for the Technical Subject Area PCS, PCN, and Mobile Systems and Networks Personal Communications Services: (Greg Pollini).

Comparison of some algorithms for tap weight evaluation in adaptive echo cancellers

The performance of some adaptive algorithms for coefficient updating of a digital echo canceller are compared. The examined algorithms are the least mean square one (LMS), the normalized LMS and the simplified versions employing the sign information. Their performance are evaluated on the basis of the echo return loss enhancement (ERLE) steady state value and convergence speed. For the algorithms employing a linear function of the error, as gradient estimate, the resuIts show that convergence speed is dependent on the echo canceller tap number and that its trend is exponential. Algorithms employing the error sign as gradient estimate are the slowest if the same variance of residual echo must be obtained. Furthermore some consideration are made in comparison with an algorithm designed in order to minimize the mean square error evaluated over a M sample block.