Wideband Speech Research Articles

Impairment Factor Framework for Wide-Band Speech Codecs

A new method is described for quantifying the quality degradation introduced by wide-band speech codecs via a one-dimensional impairment factor. The method is based on auditory listening-only tests, but the resulting impairment factors may be used for predicting speech quality in an instrumental way, e.g., for network planning purposes. Following the method, auditory test results are first transformed to an overall quality rating scale, and then adjusted to rule out test-specific effects. The derived impairment factors fit into the common framework which is defined by the E-model for narrow-band telephone networks, and which is hereby extended towards wide-band speech transmission. This paper presents the necessary auditory test data, describes the derivation and adjustment methodology, and provides numerical values for a range of wide-band speech codecs. The values are tested for their robustness in case of codec tandems and adjusted to represent the effects of packet loss

A Multiresolution Model of Auditory Excitation Pattern and Its Application to Objective Evaluation of Perceived Speech Quality

This paper proposes a multiresolution model of auditory excitation pattern and applies it to the problem of objective evaluation of subjective wideband speech quality. The model uses wavelet packet transform for time-frequency decomposition of the input signal. The selection of the wavelet packet tree is based on an optimality criterion formulated to minimize a cost function based on the critical band structure. The models of the different auditory phenomena are reformulated for the multiresolution framework. This includes the proposition of duration dependent outer and middle ear weighting, multiresolution spectral spreading, and multiresolution temporal smearing. As an application, the excitation pattern is used to define an objective measure of auditory distortion of a distorted speech signal compared to the undistorted one. The performance of this objective measure is evaluated with a database of various kinds of NOISEX-92 degraded wideband speech signals in predicting the subjective mean opinion score (MOS) and is compared with the fast Fourier transform (FFT)-based ITU-T PESQ P.862.2 algorithm. The proposed measure is found to achieve comparable correlation between subjective MOS and objective MOS as PESQ P.862.2, with a trend suggesting better correlation for the nonstationary degradations compared to the stationary ones. Further refinement of the measure for distortion types other than additive noise is anticipated

A Novel Audio Coding Scheme Using Warped Linear Prediction Model and the Discrete Wavelet Transform

In this paper, we present a novel audio coder using the discrete wavelet transform (DWT) and warped linear prediction (WLP). In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory system. The structure of the system is similar to the transform coded excitation techniques used in wideband speech coding, where LP has been replaced with WLP, and the residual is analyzed by a wavelet filterbank designed to approximate the critical bands. The inherent shaping of the WLP synthesis filter, and a controlled bit allocation to the wavelet coefficients helps minimise the perceptually significant noise due to the quantization error in the residual. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent to transparent quality for a variety of speech and music signals at an average bitrate of about 64 kb/s. Tests also show that the coder (in its initial implementation) delivers superior quality to the MPEG layer III and comparable quality to the MPEG2-AAC codec when operating at the same bitrate

Transient detection algorithms for speech coding applications

One of the methods for improving transmitted speech quality is efficient encoding of transients. The majority of currently used speech codecs employ the parametric approach, thus they do not differentiate between transients and other signal components. A sine+noise+transients signal model, proposed earlier as a tool for wideband audio signal encoding, seems to be a promising solution for enhancing signal quality in speech coding. Methods of transient identifying for an efficient transform‐domain encoding of transients has to be chosen carefully. Different transient detection algorithms are described in literature, most of them optimized for musical signals. The methods do not ensure efficient transient detection in speech signal, due to the different character of musical signals and speech. This paper provides a comparison of several transient detection methods, mostly based on evaluation of signal energy, that are now primarily used for musical signals. Their efficiency and application to transient detection in speech are evaluated and discussed. Differences between transient detection in wideband musical signals and speech are pointed out. Modifications to existing algorithms are proposed, allowing efficient transient detection and encoding in the proposed, wideband speech codec. [Research funded by the Polish Ministry of Education and Science within Grant No. 3 T11D 004 28.]

Empirical Lower Bound on the Bitrate for the Transparent Memoryless Coding of Wideband LPC Parameters

In this letter, we determine empirical lower bounds on the bitrate required to transparently code linear predictive coding (LPC) parameters derived from wideband speech. This is achieved via extrapolation of the operating distortion-rate curve of an unconstrained vector quantizer that is trained using artificial vectors generated by a Gaussian mixture model. Memoryless coding is considered and two competing LPC parameter representations are investigated. Our results show a lower bound of 31 bits/frame when assuming high-rate linearity in the operating distortion-rate curve and 35 bits/frame for an exponential curve. We also evaluate a recent quantization scheme and compare its performance against this lower bound

A G.711 Embedded Wideband Speech Coding for VoIP Conferences

This paper proposes a wideband speech coder in which a G.711 bitstream is embedded. This coder has an advantage over conventional coders in that it has a high interoperability with existing terminals so costly transcoding involving decoding and re-encoding can be avoided. We also propose a partial mixing method that effectively reduces the mixing complexity in multiple-point remote conferences. To reduce the complexity, we take advantage of the scalable structure of the bitstream and mix only the lower band of the signal. For the higher band, the main speaker location is selected among remote locations and is redistributed with the mixed lower-band signal. By subjective evaluations, we show that the speech quality can be maintained even when the speech signals are partially mixed.

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications

Standardization of the AMR wideband speech codec in 3GPP and ITU-T

3GPP and ITU-T have standardized a multi-rate codec for wideband speech conversational applications. Following a competitive selection process, the adaptive multirate wideband (AMR-WB) specifications were approved in March 2001 as part of 3GPP release 5. The ITU-T Study group 16 approved the same wideband coding algorithm as Recommendation G.722.2 and its Annexes

On the architecture, operation, and applications of VMR-WB: the new cdma2000 wideband speech coding standard

This article is an overview of the architecture and operation of the VMR-WB5 a source- and network-controlled variable-rate multimode codec designed for robust processing of wideband speech. To enable a smooth transition from legacy narrowband voice services, VMR-WB is also capable of processing conventional telephone-bandwidth speech. The VMR-WB codec is interoperable with AMR-WB at certain bit rates, thus eliminating quality degradation and additional delay due to transcoding

Noise robust bandwidth extension of telephone speech for mobile and landline communications

The present day telephone infrastructure limits the bandwidth of telephone speech to 300–3400 Hz, whereby conversational speech falls mostly within 50–8000 Hz. The ensuing band-limited narrow-band (NB) speech has a considerable loss in sound quality, in terms of naturalness and intelligibility. The purpose of bandwidth extension (BWE) is to utilize the receiver end NB speech and to add relevant frequency information to the missing low and high frequencies. This novel BWE algorithm utilizes the linear source filter model, which separates NB speech into two roughly independent components, the excitation and the spectral envelope. These components are then extended and convolved together to attain reconstructed wideband (RWB) speech. Excitation extension involves harmonic reconstruction for voiced frames and some modulations and filtering steps for unvoiced frames. Spectral envelope extension utilizes a Gaussian mixture model trained via the expectation maximization algorithm, which is involved in the mapping of NB noise robust features to the corresponding RWB features. Noise analysis is used to isolate the best feature set for real world applications. Results from a subjective listening test indicate this algorithm has an improvement in sound quality, while having the lowest known distortion metric among other speaker independent bandwidth extension algorithms.

Performance characterization of VMR-WB, the cdma2000 wideband speech codec

The voice services sub-working group of 3GPP2 conducted a standardization effort to select and characterize the first wideband speech codec for cdma2000reg. This article describes the selection process, the results of the selection process, and the subsequent tests to characterize the performance of the selected wideband speech codec, VMR-WB

Wideband speech coding standards and wireless services [Guest editoral

Wideband speech coding standards and wireless services [Guest editoral

Bandwidth extension of speech signals: a catalyst for the introduction of wideband speech coding?

The restricted audio quality of today's telephone networks is mainly due to the narrowband (NB) limitation to the frequency range from about 300 Hz to 3.4 kHz. Meanwhile, codecs for wideband (WB) telephony (50 Hz to 7 kHz) exist with significantly improved speech intelligibility and naturalness. However, the broad introduction of wideband speech coding requires strong efforts of both network operators and their customers because many elements of the networks (i.e., terminals and network nodes) have to be modified. An intermediate step to overcome the narrowband limitation can be achieved by applying artificial bandwidth extension (BWE) in the receiver. In this article we review the basic principles of bandwidth extension, and discuss several application scenarios in which both wideband coding and BWE complement each other. The introduction of BWE methods in terminals and networks may help to speed up the introduction of true wideband speech coding in the near future

Low-Complexity Source Coding Using Gaussian Mixture Models, Lattice Vector Quantization, and Recursive Coding with Application to Speech Spectrum Quantization

In this paper, we use the Gaussian mixture model (GMM) based multidimensional companding quantization framework to develop two important quantization schemes. In the first scheme, the scalar quantization in the companding framework is replaced by more efficient lattice vector quantization. Low-complexity lattice pruning and quantization schemes are provided for the $E_8$ Gossett lattice. At moderate to high bit rates, the proposed scheme recovers much of the space-filling loss due to the product vector quantizers (PVQ) employed in earlier work, and thereby, provides improved performance with a marginal increase in complexity. In the second scheme, we generalize the compression framework to accommodate recursive coding. In this approach, the joint probability density function (PDF) of the parameter vectors of successive source frames is modeled using a GMM. The conditional density of the parameter vector of the current source frame based on the quantized values of the parameter vector of the previous source frames is used to generate a new codebook for every current source frame. We demonstrate the efficacy of the proposed schemes in the application of speech spectrum quantization. The proposed scheme is shown to provide superior performance with moderate increase in complexity when compared with conventional one-step linear prediction based compression schemes for both narrow-band and wide-band speech.

Low-complexity feature-mapped speech bandwidth extension

Today's telecommunications systems use a limited audio signal bandwidth. A typical bandwidth is 0.3-3.4 kHz, but recently it has been suggested that mobile phone networks will facilitate an audio signal bandwidth of 50 Hz-7 kHz. This is suggested since an increased bandwidth will increase the sound quality of the speech signals. Since only few telephones initially will have this facility, a method extending the conventional narrow frequency-band speech signal into a wide-band speech signal utilizing the receiving telephone only is suggested. This will give the impression of a wide-band speech signal. The proposed speech bandwidth extension method is based on models of speech acoustics and fundamentals of human hearing. The extension maps each speech feature separately. Care has been taken to deal with implementation aspects, such as noisy speech signals, speech signal delays, computational complexity, and processing memory usage.

Influence of Different Speech Representations and HMM Training Strategies on ASR Performance

This work studies the influence of various speech signal representations and speaking styles on the performance of automatic speech recognition (ASR). The efficiency of two approaches to hidden Markov model (HMM) training are compared.Common MFCC and PLP features were exposed to two sources of disturbance applied to the original wide-band speech: (i) stress (Lombard effect) and (ii) transfer channel distortion (simulated telephone line). Subsequently, the efficiencies of the two training strategies were evaluated. Finally, a study of the optimal number of training iterations is introduced.

On the perceptual performance limitations of echo cancellers in wideband telephony

In this paper, standard echo canceller performance measures are evaluated in terms of psychoacoustic aspects of human hearing. The focus is on wideband speech communications systems with long round-trip delays of 200 ms and up present in the transmission path. The results of a simple acoustic echo cancellation experiment are analyzed with a standard psychoacoustic model, revealing that steady-state echo return loss enhancement and mean square error cannot be used to determine whether residual echo is perceivable in the presence of background noise. In addition, a simple modification to the normalized least mean square (NLMS) algorithm is introduced by adding a perceptual preemphasis filter. Simulation results and listening tests show that it is possible to improve the perceived performance of an echo canceller during convergence by placing greater emphasis on frequencies at which the human auditory system is most sensitive.

A comparative study of LPC parameter representations and quantisation schemes for wideband speech coding

In this paper, we provide a review of LPC parameter quantisation for wideband speech coding as well as evaluate our contributions, namely the switched split vector quantiser (SSVQ) and multi-frame GMM-based block quantiser. We also compare the performance of various quantisation schemes on the two popular LPC parameter representations: line spectral frequencies (LSFs) and immittance spectral pairs (ISPs). Our experimental results indicate that ISPs are superior to LSFs by 1 bit/frame in independent quantiser schemes, such as scalar quantisers; while LSFs are the superior representation for joint vector quantiser schemes. We also derive informal lower bounds, 35 and 36 bits/frame, for the transparent coding of LSFs and ISPs, respectively, via the extrapolation of the operating distortion-rate curve of the unconstrained vector quantiser. Finally, we report and discuss the results of applying the SSVQ with dynamically-weighted distance measure and the multi-frame GMM-based block quantiser, which achieve transparent coding at 42 and 37 bits/frame, respectively, for LSFs. ISPs were found to be inferior to the LSFs by 1 bit/frame. In our comparative study, other quantisation schemes that were investigated include PDF-optimised scalar quantisers, the memoryless Gaussian mixture model-based block quantiser, the split vector quantiser, and the split-multistage vector quantiser with MA predictor from the AMR-WB (ITU-T G.722.2) speech coder.

Multiband Vector Quantization Based on Inner Product for Wideband Speech Coding

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16 kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.

Signal modification method for variable bit rate wide-band speech coding

This paper introduces a novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates. The method is enabled only in stable voiced speech frames, and preserves the original time scale at the end of each frame. This feature helps to avoid artifacts and simplifies an encoder implementation. The signal modification includes a classification algorithm as an integral part. The classification algorithm detects the frames most suitable for signal modification and low bit rate coding, and can be employed in a rate selection module of variable bit rate (VBR) codecs. In this paper, the signal modification method is applied in an experimental VBR wide-band speech codec derived from the 3GPP adaptive multirate wideband (AMR-WB) standard (ITU-T Recommendation G.722.2). The codec fulfills the system requirements of IS-95/CDMA2000 Rate Set II, operating at source coding bit rates 12.65, 6.2, and 1.0 kb/s. The signal modification is used in the 6.2 kb/s mode dedicated for voiced speech frames. Listening test results demonstrate the good performance of the proposed method. The signal modification method is used in the Nokia/VoiceAge codec that was declared in April 2003 as the winner of the selection phase in the 3GPP2 CDMA2000 wide-band speech codec standardization.