Time-scale Modification Of Speech Research Articles

Parameters evaluation of SOLA algorithm for time scale modification

Successful operation of the Synchronous Overlap and Add (SOLA) algorithm for Time Scale Modification (TSM) of speech is closely tied to the proper choice of parameters. This paper investigates the quality of time scale modified speech under different values of primary parameters. Based on Mean Opinion Score (MOS) tests and Bark Spectral Distortion (BSD) measure, the proper choices of synthesis shift (Ss) and the duration of the shift search interval (K max ) are given experimentally. The conclusions can be helpful for operating the SOLA algorithm for time scale modification of speech.

Adaptive time scale modification of speech for graceful degrading voice quality in congested networks for VoIP applications

This paper proposes an alternative scheme to variable bit rate (VBR) speech coding for voice over Internet protocol (VoIP) during network congestion in Internet. The proposed scheme is called “adaptive bit rate switching” and ensures that the available bandwidth is most efficiently used. When congestion is signaled, a time scale modification algorithm called WSOLA (waveform similarity overlap and add) with time-dependent compression rate, determined according to the severity of the network congestion, is employed in order to reduce the bit rate required to transmit speech adaptively. This approach is different from VBR speech coding and novel in the sense that the coder operates at any desired bit rate for any desired duration. This is particularly useful in network environments because load may be different at each direction. WSOLA algorithm has been selected as the time scale modification algorithm because it is computationally efficient and produces high quality output. In addition, the proposed scheme integrates WSOLA, or any time scale modification algorithm into any commercial or military constant bit rate (CBR) or VBR codec without any modification in the vocoder structure. The results of the proposed method are statistically evaluated by using diagnostics rhyme tests (DRT) and mean opinion score (MOS) tests. The DRT results obtained from the simulation of the proposed system revealed, under 90% confidence interval, that the perceptual success of the adaptively compressed and G.729 coded speech is 98.92±0.03 percent. The MOS test results, on the other hand, proved that the system provides better perceptual quality than the standard time scale modification, indicating that the proposed system indeed provides graceful degradation in voice quality even in additive increase multiplicative decrease modeled channels, provided that the dynamic network conditions grant bandwidth.

Concatenative synthesis based on a harmonic model

One of the most successful approaches to synthesizing speech, concatenative synthesis, combines recorded speech units to build full utterances. However, the prosody of the stored units is often not consistent with that of the target utterance and must be altered. Furthermore, several types of mismatch can occur at unit boundaries and must be smoothed. Thus, both pitch and time-scale modification techniques as well as smoothing algorithms play a crucial role in such concatenation based systems. In this paper, we describe novel approaches to each of these issues. First, we present a conceptually simple technique for pitch and time-scale modification of speech. Our method is based upon a harmonic coding of each speech frame, and operates entirely within the original sinusoidal model. Crucially, it makes no use of "pitch pulse onset times." Instead, phase coherence, and thus shape invariance, is ensured by exploiting the harmonic relation existing between the sine waves used to code each analysis frame so that their phases at each synthesis frame boundary are consistent with those derived during analysis. Secondly, a smoothing algorithm, aimed specifically at correcting phase mismatches at unit boundaries, is described. Results are presented showing our prosodic modification techniques to be highly suitable for use within a concatenative speech synthesizer.

Non-parametric techniques for pitch-scale and time-scale modification of speech

Time-scale and, to a lesser extent, pitch-scale modifications of speech and audio signals are the subject of major theoretical and practical interest. Applications are numerous, including, to name but a few, text-to-speech synthesis (based on acoustical unit concatenation), transformation of voice characteristics, foreign language learning but also audio monitoring or film/soundtrack post-synchronization. To fulfill the need for high-quality time and pitch-scaling, a number of algorithms have been proposed recently, along with their real-time implementation, sometimes for very inexpensive hardware. It appears that most of these algorithms can be viewed as slight variations of a small number of basic schemes. This contribution reviews frequency-domain algorithms (phase-vocoder) and time-domain algorithms (Time-Domain Pitch-Synchronous Overlap/Add and the like) in the same framework. More recent variations of these schemes are also presented.

Effect of time scale modification of speech on the speech recognition threshold in noise for elderly listeners.

The effect of time scale modification of speech on the speech recognition threshold in noise (SRTN) was investigated for a group of 44 elderly subjects, varying in age from 56 to 88 years with sloping mild-to-moderate sensorineural hearing losses. Subjects' pure-tone average thresholds at 500, 1,000 and 2,000 Hz (PTA1) ranged from 3 to 58 dB. The study confirms the well-known fact that speech recognition in noise deteriorates with increasing age. Furthermore, time-compressing speech results in an even greater influence of age on the SRTN. The apparent age effect on SRTN values referenced to SRTN in the control condition is mostly caused indirectly by the significant correlation between age and PTA1, although there was a slight tendency for age to have a greater independent effect on relative SRTN values when the amount of time compression increased. The percentage of explained variance in relative SRTN values by the subject variables in this study was limited, indicating that there are other factors not investigated in this study (e.g. cochlear mechanisms, short-term memory, feature extractors) that significantly influence the effect of time compression on SRTN.