Modulated Noise Reference Unit Research Articles

Time-frequency voiced and unvoiced excitation models for harmonic speech systems

Time-frequency voiced and unvoiced models are proposed for the excitation of a harmonic autoregressive wideband speech analysis-synthesis system. The time-frequency voiced excitation (TFVEX) model has low time resolution defined by the concentration of the excitation signal distribution in the modulation domain while the time-frequency unvoiced excitation (TFUNEX) model has cycle time discrimination with lower amplitude resolution and while the frequency resolution for both models is an octave. The speech reconstructed by the compound TFUVEX unvoiced-voiced model is rated above the speech degraded by a modulated noise reference unit (MNRU) at 25 dB in listening tests while yielding a parametric compression of over ten times.

Time-frequency voiced and unvoiced excitation models for harmonic speech systems DOI: 10.14209/jcis.2014.6

Time-frequency voiced and unvoiced models are proposed for the excitation of a harmonic autoregressive wideband speech analysis-synthesis system. The time-frequency voiced excitation (TFVEX) model has low time resolution defined by the concentration of the excitation signal distribution in the modulation domain while the time-frequency unvoiced excitation (TFUNEX) model has cycle time discrimination with lower amplitude resolution and while the frequency resolution for both models is an octave. The speech reconstructed by the compound TFUVEX unvoiced-voiced model is rated above the speech degraded by a modulated noise reference unit (MNRU) at 25 dB in listening tests while yielding a parametric compression of over ten times.

Voice quality measurement in telecommunication networks by optimized multi-sine signals

The paper deals with the selection of the test signal for the voice quality measurement (VQM) in telecommunication networks. It lays the fundamentals to use the optimized multi-sine as test signal in place of both the natural and the artificial voices, suggested by the ITU-T recommendations. The effectiveness of the optimized multi-sine to the VQM is investigated by using the four algorithms: (i) perceptual speech quality measurement, (ii) perceptual speech quality measurement plus, (iii) perceptual evaluation of speech quality, and (iv) perceptual analysis measurement system. Tests are performed in order to compare the mean output index of the four algorithms feed by the optimized multi-sine, the natural and the artificial voice signals, respectively. In order to take into account the effects of the actual telecommunication networks, different characteriztic noise is used to corrupt both the multi-sine and the voices: (i) modulated noise reference unit, (ii) impulsive noise, and (iii) additive white Gaussian noise. Further tests are performed by simulating both the transmission and the reception processing of the GSM standard. The tests show the effectiveness of the technique based on the optimized multi-sine to the VQM with respect to the traditional one.

Continuous assessment of time-varying speech quality.

This paper addresses the question of whether subjects are able to assess the perceived time-varying quality of speech material continuously. A method is introduced which is characterized by a subjective continuous rating of the perceived speech quality by moving a slider along a graphical scale. The usability of this method is illustrated with an experiment in which different sequences of sentences were degraded in quality with a Modulated Noise Reference Unit. The modulation depth was varied with time and the subject's task was to assess the perceived quality. The results indicate that subjects can monitor speech quality variations very accurately with a delay of approximately 1 s. An objective speech quality measure based on an auditory processing model was applied to predict the subjective speech quality results. The speech quality measure qC was modified to allow for time-dependent objective measurement of the speech quality. The averaged subjective response data could be modeled by the scale transformed and low-pass filtered measure qC(t) with a high degree of accuracy.

Modeling speech intelligibility and quality on the basis of the ‘‘effective’’ signal processing in the auditory system

A model of the ‘‘effective’’ signal processing performed by the auditory system is applied to the quantitative assessment of speech intelligibility (cf., Kollmeier et al., Proc. ICA 95, III, 81–84). It consists of a gamma one filterbank followed by envelope extraction, nonlinear adaptation circuits, and subsequent modulation low-pass filtering in each filter channel before being fed into a cross-correlation-type detector or pattern recognizer. The model is capable of predicting the speech intelligibility of both normal and hearing-impaired listeners under a wide range of signal-to-noise ratios, as well as the perceived quality of speech signals whose spectro-temporal properties are systematically degraded. The degradations are generated with a modulated noise reference unit either in a band-specific, time-independent way or in a broadband, temporally varying way. These intelligibility and quality assessment data can be used to deduce the components of the model most germane to the processing of speech, as well as provide information about how to enhance and extend the model for better predictive performance. a)Now at Siemens Audiologische Technik, Erlangen, Germany. b)Now at Widex Hearing Aids, Vaerloese, Denmark.

Continuous assessment and modeling of speech transmission quality

This paper addresses the question of whether subjects are able to assess the perceived time-varying quality of speech material continuously. A method inspired by Hamberg and de Ridder [J. Opt. Soc. Am. A 12, 2573–2577 (1995)] is presented by which subjects continuously rate the perceived speech quality by moving a slider along a graphical scale. In this way, temporal variations in quality can be monitored quantitatively and differences between, for example, alternative speech transmission systems, can be analyzed in an informative way. The usability of this method is illustrated with an experiment in which different sequences of sentences with a total duration of 40 s were degraded in quality with a modulated noise reference unit. The modulation depth was varied with time and the subjects task was to assess the perceived quality. The results show that subjects can monitor speech quality variations closely with a delay of approximately 1 s. The subjective results are discussed in view of a psychoacoustically motivated objective speech quality measure [Hansen and Kollmeier, ICASSP ’97, 1387–1390 (1997)].

8 kbit/s short and medium delay speech codecs based on CELP coding

AbstractThis paper discusses two speech‐coding algorithms at around 8 kbit/s. These algorithms both use pre‐selection in the codebook search and training for the conjugate structured random codebook. One has short‐delay time that can dispense with echo‐control equipment. This algorithm is robust against channel errors by using LPC analysis separated from pitch excitation, although LPC parameters are derived from the locally decoded signal. The other algorithm has medium‐delay time and meets all the requirements of the ITU‐T (formerly CCITT) 8 kbit/s speech coding standard. The performance of these codecs is evaluated using signal to noise ratio (SNR), pair comparison tests with a modulated noise reference unit (MNRU), and Mean Opinion Score (MOS). The major applications of these codecs are personal handy phone systems and FPLMTS.

Low delay tree coding of speech at 8 kbit/s

A low delay tree coder for coding speech at 8 kbit/s that incorporates a new backward coefficient adaptation structure is presented. The coder has an algorithmic delay of 2.5 ms, produces good quality speech over ideal channels, and maintains robustness to bit errors for independent bit error rates up to and including 10/sup -2/. The tree coder consists of a randomly populated fractional rate tree, a weighted squared error distortion measure, the (M,l) tree search algorithm, an incremental path map symbol release rule, a long-term predictor, an all-pole short-term predictor, and the newly-obtained parameter adaptation algorithms; The sensitivity to channel errors is reduced by using the receiver excitation sequence for adaptation of the short-term predictor coefficients, but the ability to track rapid changes in the speech is retained by shaping the excitation sequence with an all-zero filter. The output speech of the tree coder is evaluated using segmental signal-to-noise ratios (SNRSEG), narrow band spectrograms, a paired comparison with pulse code modulation (PCM), and an equivalent noise paired comparison using modulated noise reference unit (MNRU).< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Effect of generation method of modulated-noise reference signal on speech quality evaluation

This paper discusses the modulated-noise reference unit (MNRU), which is used to represent quantitatively the effect of the quantization distortion due to voice coding on the voice quality. The noise generation method, which has only implicitly been discussed, is formulated; and the effect of various generation parameters on the voice quality evaluation is examined. The following results are obtained. The noise source has a uniform amplitude distribution which simplifies the digital processing compared to the generation of noise with a Gaussian distribution. The effect on the quality is small even if the noise-generation cycle is short. The simplified method proposed by Bellcore Co., U.S.A., is examined for comparison. It is shown through an international evaluation experiment that there is no significant difference in the result of quality measurement even if the simplified generation method is adopted.

Comparison of three methods for the subjective measurement of speech quality.

Three methods for the subjective measurement of speech quality, namely, the Diagnostic Acceptability Measure (DAM), the Absolute Category Rating method (ACR), and the Degradation Category Rating method (DCR), were used to evaluate a set of 24 systems conditions using a panel of 20 listeners. The DAM yields a Composite Acceptability Estimate (CAE), the ACR a Mean Opinion Score (MOS), and the DCR a Degradation Mean Opinion Score (DMOS). The 24 systems included three subsets: 12 modulated noise reference unit (MNRU) conditions, six speech to noise ratio (S/N) conditions, and six ‘‘n-bit quantization’’ conditions. All three methods provided scores that were monotonic over the range of values evaluated for each of the three subsets of system conditions. Analyses of variance, however, indicated differential resolving power for the three methods, across each subset of system conditions. Further analyses, including transformation of scores to a common scale, showed a linear relationship between CAE and MOS, but curvilinear relationships between CAE and DMOS and between MOS and DMOS.

Modulated noise reference unit (MNRU) tests using DRT, DAM, and spelling alphabet test materials

The modulated noise reference unit (MNRU) has been widely used for reference conditions in evaluating voice coders for telephone applications. Opinion scores using the mean opinion score (MOS) are referenced to equivalent modulated signal‐to‐noise level. The DoD Digital Voice Processor Consortium has conducted extensive tests of voice processors using the Diagnostic Rhyme Test (DRT) to evaluate intelligibility and the Diagnostic Acceptability Measure (DAM) to evaluate acceptability. Spoken materials for the DRT, the DAM, and the ICAO spelling alphabet words and digits were degraded by speech‐modulated noise at levels from + 30 to − 15 dB. The resulting tapes were evaluated by trained listeners. Even small amounts of noise degraded acceptability, DRT intelligibility degraded more slowly, and the spelling alphabet was moderately intelligible even at the worst speech to noise levels. Relations among the different types of tests will be discussed.