Toll Quality Research Articles

A novel approach for artificial bandwidth extension of speech signals by LPC technique over proposed GSM FR NB coder using high band feature extraction and various extension of excitation methods

This research addresses an issue of wide band (WB) speech transmission (having cut-off frequency $$\hbox {f}_{\mathrm{c}}=8$$fc=8 kHz) over standard narrow band (NB) communication link (supporting bandwidth of 300---3,400 Hz). A long transition time for technological up-gradation from NB to WB systems eventually lead to development of backward compatible techniques such as artificial bandwidth extension (ABE) which is capable of providing bandwidth of 50---7,000 Hz, in turn contributing toll quality recovered speech at receiving end. This paper investigates a novel approach to compute high band (HB) features using linear predictive coding (LPC) technique at transmitter from given input WB speech corpus. These encoded features are embedded into bitstream of proposed GSM Full Rate 06.10 NB speech coder using joint source coding and data hiding technique and then transmitted to receiver. At receiver, these HB features are extracted to reproduce HB recovered speech using watermark extraction algorithm and for the same different extension of excitation techniques have been adopted and implemented. An e-test bench is created to implement this proposed ABE coder in MATLAB and series of simulations are carried out using Subjective (mean opinion score--MOS) and Objective (perceptual evaluation of speech quality--PESQ) analysis. Obtained results for both analyses advocate performance improvement of proposed ABE coder over legacy GSM 06.10 FR NB coder for various extension of excitation techniques.

Substitution of G.728 vocoder’s codebook search module with SOM array trained by PSO-optimized supervised algorithm

Low delay-code excited linear prediction (LD-CELP) is an attractive algorithm in implementing vocoders in voice over Internet protocol networks. This algorithm has been proposed for the coding of speech at 16 kbps with toll quality. However, operation at transmission rates lower than 16 kbps is desirable, so that traffic can be accommodated during system overload conditions. In this paper, an array of self-organizing maps (SOMs) is employed instead of traditional codebook search module, recommended in ITU-T G.728, to determine the optimum index value of shape codebook. It is noted that a modified supervised training algorithm is used for SOMs in which some of the training parameters are optimized using particle swarm optimization (PSO) algorithm. Based on the occurrence frequency characteristics of codevectors, six bits for shape codebook and two bits for gain codebook are used in this work to produce a vocoder with lower bit rate as compared with traditional ITU-T G.728 vocoder. The performance comparison of the proposed SOM array trained by PSO-optimized supervised algorithm as the codebook search module in the structure of LD-CELP with a conventional implementation of LD-CELP coder shows that execution time of the algorithm is reduced up to 44 %. However, the degradation of voice quality in terms of mean opinion score, perceived evaluation of speech quality and segmental signal-to-noise ratio (SNRseg) is acceptable.

Subjective audio quality over a secure IEEE 802.11n network

This paper presents an empirical evaluation of audio quality generated by a G.711 codec and transmitted over IEEE 802.11n, IEEE 802.11b, and IEEE 802.11g Wireless Local Area Networks (WLANs). Audio quality decline due to additional calls or by securing the WLAN with Internet Protocol Security (IPsec) is quantified. Results suggest that audio quality over an IEEE 802.11n WLAN is not higher than over an IEEE 802.11b WLAN for up to 10 simultaneous calls. The data strongly suggest that toll quality audio (MOS ≥ 4.0) is not currently practical over IEEE 802.11 WLANs secured with WPA2, even using the G.711 codec.

Low bit rate speech coding

This paper describes the objective of speech coding to achieve toll quality performance at minimum bit rate and to improve the efficiency of transmission and storage, reduce cost, increase security and robustness in transmission. Speech coding can be achieved based on two facts: redundancy in speech signals, and the perception properties of human ears. Performance evaluation of speech coding is obtained by speech quality, coding rate, algorithm complexity and delay. The two speech coding methods: waveform coding and parametric coding uses a synthetic model for speech analysis and reconstruction. This paper focuses on a source coding and subjective mean opinion score methods are used to measure the speech perception based on score given by the listener to achieve low bit rate coding.

Hybrid multimode/multirate CS-ACELP speech coding for adaptive voice over IP

This paper proposes a variable rate, toll quality CS-ACELP coder that uses coding modes compatible with the three 6.4, 8 and 11.8 kbit/s coding schemes standardised by ITU-T in G.729. In particular we propose a hybrid multimode/multirate (M 3R) codec presenting four coding categories, with an average bit rate ranging between 3 and 8 kbit/s, that adapts the rate to changes in network conditions. The results indicate that both for the current Internet and for future scenarios hybrid M 3R adaptive coding seems a valid solution to achieve the quality currently guaranteed by PSTN networks.

CorDECT—A Mature and Effective Alternative to the Wireline Local Loop

CorDECT is an advanced, low-cost wireless access system (WAS) that provides toll quality voice and simultaneous Internet access at 35/70 kbps. The system is designed to be deployed in both dense, urban and sparse, rural areas.

Low complexity speech coding at 1.2 to 2.4 kbps based on waveform interpolation

Low-rate speech coding technology has recently made a significant progress with the introduction of new interpolative algorithms (Shoham, 1993; Kleijn and Haagen, 1995). The inherent complexity of these algorithms is, however too high too be commercially useful for low-cost applications. In this paper we propose new approaches to low-complexity speech coding at coding rates of 1.2 and 2.4 kbps. The proposed methods utilize all the advantages of interpolatve coding but greatly simplify the analysis and synthesis operations to a point where low-cost two-way digital speech communication can be easily implemented on DSP or host platforms. At 2.4 kbps, the complexity of the proposed coder is about 7.5 and 2.5 MFLOPS for the encoder and decoder, respectively. At 1.2 kbps, the complexity is about 6 and 2.3 MFLOPS for the encoder and decoder, respectively. The small computational load of these coders make them suitable for multi-tasking environment and low-cost terminals. Informal subjective evaluation shows that, at 2.4 kbps, good communication quality is obtained. Communication quality is less than toll quality but the perceived coding effects are not annoying and do not prevent long sustained two-way conversation with high degree of intelligibility. The quality does not significantly degrade at 1.2 kbps and it is considered sufficient for messaging applications.

The Enhanced Variable Rate Coder: Toll quality speech for CDMA

The Enhanced Variable Rate Coder (EVRC), standardized by the Telecommunications Industry Association (TIA) as IS-127, is intended for use with the IS-95x Rate Set 1 air interface (CDMA). This coder operates at a maximum rate of 8.5 kb/s and an average rate of about 4.1 kb/s on conversational speech. The EVRC consists of three coding modes that are all based on the Code Excited Linear Prediction (CELP) model. Selection among the three modes is based on an estimate of the input signal state, with active speech encoded primarily at 170 bits/20 msec frame (Rate 1), background noise and silence encoded at 16 bits/frame (Rate 1/8), and some active speech and essentially all transitions between speech and silence encoded at 80 bits/frame (Rate 1/2). In order to improve performance in the presence of background noise, the EVRC employs an adaptive noise-suppression filter at the input. Subjective test results are presented demonstrating that the EVRC delivers excellent quality voice in clean speech/clear channel conditions, and that its performance exceeds that of most currently standardized speech coders for wireless applications in background noise and/or impaired channel conditions.

Speech-and-channel coding scheme for enhancedfull-rate GSM standard

The enhanced full-rate (EFR) GSM speech channel operates at the same gross data rate as the existing full-rate channel, TCH-FS. It is an upgrade for the latter in both speech quality and channel robustness. In this investigation. A speech and channel coding scheme is devised to produce almost toll quality speech even in severe channel conditions.

A toll quality 8 kb/s speech codec for the personal communications system (PCS)

A toll quality speech codec at 8 kb/s suitable for the future personal communications system is presented. The codec is currently under standardization by the ITU-T (successor of CCITT) where the codec terms of reference were mainly determined considering PCS application. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP) and has a speech frame of 10 ms. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Formal subjective listening tests, performed by ITU-T SG 12, showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kb/s in error-free conditions and it outperforms G.726 under error conditions. The codec performs adequately under tandeming conditions, and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements, and this is one subject of study for the next phase. The algorithm has been implemented on a single fixed-point DSP for the ITU-T subjective rest, and required about 29 MIPS. An optimized version, however, requires 24 MIPS without any speech quality degradation.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Non-voice performance of the 16 kbit/s LD-CELP algorithm

Recently, CCM adopted a Low-Delay Code Excited Linear Predictive (LD-CELP) algorithm as a new recommendation for the coding of telephone bandwidth speech at 16 kbit/s with “toll quality”. In this paper, the performance of the LD-CELP algorithm with non-voice signals is presented and compared with 64 kbit/s PCM. From these results it can be determined that the LD-CELP, in a single encoding/decoding configuration. is acceptably transparent to network signaling as well as voiceband data at rates not exceeding 2.4 kbit/s.

Effects of PRMA on objective speech quality

When transmitting 32 kbit/s adaptive differential pulse (ADPCM) speech using Reed–Solomon error correction coding and 16 level quadrature amplitude modulation (16-QAM), our 20 slot packet reservation multiple access (PRMA) assisted cordless telecommunications (CT) scheme supported 36–38 speech users with negligible objective and subjective speech degradation. The average number of users per slot was nearly doubled due to deploying PRMA and toll quality speech was transmitted in a user bandwidth ∼ 11.6 kHz. For a channel signal-to-noise ratio (SNR) in excess of 25 dB, a Rayleigh fading channel and mobile speeds above 2 mph the speech segmental SNR degradation was less than 0.3 dB.

Variable bit rate adaptive predictive coder

An adaptive predictive coder providing almost toll quality at 16 kb/s and minimal degradation when the bit rate is lowered to 9.6 kb/s is described. The coder can operate at intermediate bit rates and can also change bit rate on a packet-by-packet basis. Variable bit rate operation is achieved through the use of switched quantization, thus eliminating the need for buffering of the output. A noise shaping filter provides flexible control of the output noise spectrum. The filter, in conjunction with an enhanced way to adapt the quantizer step size, which tries to accommodate the quantization noise feedback, accounts for the toll quality. By quantizing the residue with more than one quantizer, the effective number of bits per sample can be controlled in a deterministic way regardless of the entropy residue. The lower limit of operation is at 9.6 kb/s. Performance of the coder under random bit errors is also presented. It has been found that only at error rates of 10/sup -2/ and higher does the degradation becomes objectionable. >

Fractional rate multitree speech coding

The authors present both forward and backward adaptive speech coders that operate at 9.6, 12, and 16 kb/s using integer and fractional rate trees, weighted squared error distortion measures, the (M,L) tree search algorithm, and incremental path map symbol release. They introduce the concept of multitree source codes and illustrate how the multitree structure allows scalar quantizer-based codes and scalar adaptation rules to be used for fractional rate tree coding. With a frequency weighted distortion measure, the forward and backward adaptive multitree coders produce near toll quality speech at 16 kb/s, while the backward adaptive 9.6 kb/s multitree coder substantially outperforms adaptive predictive coding and has an encoding delay of less than 2 ms. Performance results are present in terms of unweighted and weighted signal-to-noise ratio and segmental signal-to-noise ratio, sound spectrograms, and subjective listening tests.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Improved excitation prediction and quantization in optimal amplitude multipulse coders

Improvements to the excitation prediction and quantization procedure in optimal amplitude multipulse coders are described. The new procedure successively reoptimizes the pulse amplitudes and predictor gain of the long‐delay correlation filter as each new pulse is found. Furthermore, the amplitudes and gain are requantized at each step so that the new pulse amplitudes and locations can correct for the quantization errors in the existing excitation. To perform the joint amplitude‐predictor reoptimization, nonrecursive pitch prediction structure is used in our analysis. The predictor is implemented as an adaptive vector quantizer whose codebook is populated with past excitation sequences. This structure allows for a computationally efficient closed‐loop analysis procedure. The results showed that at 9.6 kbps, the new technique achieved an average segmental SNR of over 18.5 dB. Informal subjective tests indicated that the reconstructed speech is toll quality.

A multi-rate transcoder

The authors describe a single-chip, half-duplex multirate transcoder device. The device is pin selectable as a 32-kb/s ADPCM (adaptive differential pulse code modulation) transcoder and a 16-kb/s subband coding transcoder, and it provides a 64-kb/s PCM passthrough option. The transcoder implementation is fully compatible with the 32-kb/s ADPCM standard defined by ANSI and CCITT recommendation G.721 bis. The subband coder (SBC) algorithm provides a high-quality, cost-effective medium bit-rate solution. The ADPCM and SBC implementations on the device produce toll quality speech. >

Multirate subband coding applied to digital speech interpolation

It has been shown that the output of a multirate subband coder has an embedded bit stream, i.e., accepts bit deletion and insertion for dynamic rate conversion. This property is of great importance in a digital voice communication network since it allows any overloaded node to simply flag and drop appropriate bits from the bit stream, without tandeming of encode/decode operations. This paper presents a statistical analysis of a digital speech interpolation (DSI) system using multirate subband coding. Our intent is to estimate the level of speech quality and grade of service that can be provided by such a system, given values of various system parameters. The analysis uses birth-death models for the arrival and departure of calls and talk-spurts, and considers a strategy for allocating bandwidth among the ports of the system that takes advantage of the properties of the multirate subband coder. A simulation model is employed to estimate values of certain performance parameters that are difficult to obtain analytically. Our bandwidth allocation strategy ensures that all traffic offered to the DSI system is carried, with neither blocking nor freezeout. We consider, as an example, a DSI system with a maximum concentration ratio of 4 to 1, in which 96 PCM speech channels at 64 kbits/s each are concentrated on a single 1.544 Mbit/s T1 link. The application of multirate subband coders is shown to be extremely efficient, since completely toll quality speech transmission is provided even at the full 4-to-1 compression ratio, without any freezeout. Preliminary results of informal listening tests verify the level of speech quality predicted by our analysis.

A Comparison of Digital Speech Coding Methods for Mobile Radio Systems

Subjective quality measurements on three digital speech coders, simulated with mobile radio channel transmission, were performed using the mean opinion score (MOS) method. The three speech coding methods tested were: continuously variable slope deltamodulation (CVSD) coding, adaptive predictive coding (APC), and residually excited linear predictive (RELP) coding. Several versions of each coder, with transmission rates in the range of 7.3 to 16.1 kbits/s, were simulated. Five different channel conditions, including three derived from land mobile radio field experiments, were applied to the speech coders' encoded output to study the effects. The results show that of the three coders, the CVSD coder is the most robust to channel errors, but produces reconstructed output speech of unacceptable quality. The 14.4 kbit/s RELP coder produces relatively good Output speech quality, exhibits a mild degree of robustness to mobile radio channel errors, and is slightly less complex than the APC coder. Of the three digital speech coders tested, the RELP coder appears the most suitable for use with land mobile radio. However none of the three coders was able to produce speech of telephone toll quality in a mobile radio environment.

Transmission of digital speech in highway microcells

Digital transmission of speech over highway microcellular channels is investigated and theoretical results of bit error rate (BER) as a function of channel SNR and co-channel interference are presented. MSK modulation is used and the co-channel interferer has both a carrier phase difference and data stream time offset from the wanted signal.Transmissions over Rayleigh fading and AWGN channels are studied, and the theoretical and simulation results are in close agreement. Simulations were also performed using log-PCM, ADPCM and SBC speech. It was found that for toll quality speech it was necessary to operate with a channel SNR and co-channel SIR that both exceeded 30 dB for a BER of 5×10−4. No channel coding or equalization was used. For free flowing vehicular traffic conditions where the average vehicular speed was 112 km/h, a three-cell cluster yielded a cell length of 2·13 km, assuming an allotted duplex bandwidth of 10 MHz, a channel spacing of 25 kHz and 133 mobile users per microcell. The spectral efficiency for 2% and 10% blocking was found to be 36·1 and 42·1 erlang/MHz/base station, while the corresponding erlang/MHz/km2 for a cell length of 2·13 km was 807·1 and 941·2.

Regular-pulse excitation--A novel approach to effective and efficient multipulse coding of speech

This paper describes an effective and efficient time domain speech encoding technique that has an appealing low complexity, and produces toll quality speech at rates below 16 kbits/s. The proposed coder uses linear predictive techniques to remove the short-time correlation in the speech signal. The remaining (residual) information is then modeled by a low bit rate reduced excitation sequence that, when applied to the time-varying model filter, produces a signal that is close to the reference speech signal. The procedure for finding the optimal constrained excitation signal incorporates the solution of a few strongly coupled sets of linear equations and is of moderate complexity compared to competing coding systems such as adaptive transform coding and multipulse excitation coding. The paper describes the novel coding idea and the procedure for finding the excitation sequence. We then show that the coding procedure can be considered as an optimized baseband coder with spectral folding as high-frequency regeneration technique. The effect of various analysis parameters on the quality of the reconstructed speech is investigated using both objective and subjective tests. Further, modifications of the basic algorithm, and their impact on both the quality of the reconstructed speech signal and the complexity of the encoding algorithm, are discussed. Using the generalized baseband coder formulation, we demonstrate that under reasonable assumptions concerning the weighting filter, an attractive low-complexity/high-quality coder can be obtained.