Hiromi Nagabuchi

Objective quality evaluation for low-bit-rate speech coding systems

An LPC (linear predictive coding) cepstrum distance measure (CD) is introduced as an objective measure for estimating the subjective quality of speech signals. Good correspondence between LPC CD and the subjective quality, expressed in terms of both opinion equivalent Q and mean opinion score, are shown. Good repeatability of objective quality evaluation using LPC CD is also shown. A method for generating an artificial voice signal that reflects the characteristics of real speech signals is described. The LPC CD values calculated using this artificial voice are almost the same as those calculated using real speech signals. The speaker-dependency of the coded-speech quality is shown to be an important factor in low-bit-rate speech coding. Even taking this factor into consideration, LPC CD is shown to be effective for estimating the subjective quality. >

Quality assessment of speech coding and speech synthesis systems

The concept of speech quality assessment is examined. Quality assessment methodologies for speech waveform coding, source coding, and speech synthesis by rule from the viewpoints of naturalness and intelligibility are reviewed. Both subjective and objective measures are considered.<<ETX>>

Speech coding technology for ATM networks

A type of speech coding for asynchronous transfer mode (ATM) is described. Cell processing, which improves service quality, is taken into account. Missing-cell recovery methods are discussed, and the distinctive features of missing-cell recovery methods used with low-bit-rate coding are examined. An example of the speech quality obtained using speech coding techniques in the ATM networks is described. The performance levels for increasing cell loss are compared for various speech coding methods, in combination with methods for dividing coded speech signals into cells and discarding cells. Representative feasible network applications of coding technologies are considered.<<ETX>>

A new artificial speech signal for objective quality evaluation of speech coding systems

Describes a new artificial speech signal (ASVQ: artificial speech by vector quantization technique) which reflects the average characteristics of the human voice. The ASVQ is intended for use as a test signal in the objective evaluation of speech coding system quality. To obtain the average characteristics, a very large speech data base is analysed, The ASVQ generation method which reflects the extracted average characteristics of the human voice is formulated. This method applies vector quantizing analysis to the speech data base. The LPC speech synthesis circuit is used to reproduce the average characteristics. Finally, the new artificial speech signal is compared with a human voice and the estimation accuracy of the subjective quality of speech coding systems and nonlinear distortions is evaluated. >

Artificial conversational speech signal generation method for measuring characteristics of devices operated by speech signals

A method for generating an artificial conversational speech signal as the input test signal in measuring the characteristics of devices operated by speech signals, such as speech detectors, voice switches, and echo controllers is proposed. A state transition model among talkspurt, pause, double‐talk, and mutual silence in conversational speech signals was introduced to simulate the statistical characteristics of these states in real conversational speech signals. This model assumed that the cumulative distribution of the duration time in each state is exponential. During talkspurt intervals, an artificial voice was generated using a speech synthesizer that controlled the spectrum and source characteristics independently. This artificial signal generation method can generate a signal with given characteristics by changing the values of system parameters, such as the average duration time of each state, transition probability from one state to another, and LPC (linear predictive coding) parameters that expr...

Artificial voice signal for objective quality evaluation of speech coding systems

An artificial voice signal that reflects the average characteristics of the human voice is described. This signal is intended for use as a test signal in the objective evaluation of speech coding system quality. To obtain the average characteristics, a multilingual set of speech samples is analyzed. An artificial voice generation method that reflects the extracted average characteristics of a human voice is formulated. This method applies vector quantizing analysis to the multilingual speech database. The partial correlation (PARCOR) speech synthesis circuit is used to reproduce the average characteristics. Finally, the artificial voice is compared with a human voice, and the estimation accuracy of the subjective quality of speech coding systems is evaluated.<<ETX>>

Analysis of quality factors in synthetic speech produced by rules

This paper investigates how various factors affect the quality of synthetic speech produced by rules. Using rules to synthesize speech will be an important technique for providing various telecommunication services in future intelligent networks. The quality of synthetic speech is generally measured by subjectively evaluating the speech from the viewpoint of intelligibility, or by comparing it with the quality of other types of synthetic speech. However, the development of a practical speech synthesis method for use in telecommunication networks requires an overall quality evaluation, including intelligibility and naturalness. The quality should be compared with that of natural telephone speech. To establish an overall quality evaluation method, the effects of several factors on the overall quality (expressed by MOS) of speech synthesized by several Japanese text‐to‐speech systems are quantitatively compared with the effects of using additive speech‐correlated white noise as a natural speech material. Exp...

A measure for selecting source signals used in quality assessment of coded music signals

In order to establish the quality assessment method for the wideband high-quality acoustic telecommunication, the selection of the source signal is one of the important issues. The effect of the source signal used in the quality assessment on the subjective quality must be analyzed, and the measure for the selection of the source signals must be developed based on the result of analysis. This paper considers the 20-kHz band music signal and a measure for selecting the source signal is proposed to be used in the assessment of the coding distortion, which is one of the deteriorating factors for the communication quality. The effectiveness of the measure is demonstrated. As the first step, the “noisiness of the mid-band signal,” the “flatness of spectrum envelop” and the “harmonicity of spectrum” are considered as the features that affect the subjective assessment score of the coded music signal. A measure for selecting the source signal based on those feature parameters is proposed and the parameters are optimized by the training data. The assessment experiment is conducted using the source signal recorded in the acoustic signal set (SQAM-CD) recommended by CCIR, and the effectiveness of the proposed selection measure is verified when it is applied to the unknown data. It is also shown that the performance of the assessment object is more accurately evaluated using the proposed selection measure than in the case where the source signals are selected at random.

Speech quality assessment in low‐bit‐rate speech coding taking into consideration quality variation among speakers

A method for assessing the quality of low‐bit‐rate speech coding is proposed. This method considers the speaker dependency of coded speech quality, which has not been done before. First, it is shown that coded speech quality varies with speakers more in low‐bit‐rate coding than in PCM. Next, the power of the linear prediction error of the signal of speech weighted by speech power, Gopt, is introduced as a measure for estimating quality variation among speakers. It is shown that Gopt, can be used as an effective index for selecting speakers in coded speech quality evaluation. Finally, a speech quality assessment method is proposed in which (1) test speakers are selected to cover uniformly the whole range of the Gopt, distribution; and (2) the assessment results are expressed in terms of not only the average but also the standard deviation of measured values for each speaker weighted according to the Gopt distribution characteristics. This method can assess the performance of low‐bit‐rate speech coding more...