Waveform Interpolation Research Articles

Framework and tools for the simulation and analysis of the radio emission from air showers at IceCube

The Surface Enhancement of the IceTop air-shower array will include the addition of radio antennas and scintillator panels, co-located with the existing ice-Cherenkov tanks and covering an area of about 1 km2. Together, these will increase the sensitivity of the IceCube Neutrino Observatory to the electromagnetic and muonic components of cosmic-ray-induced air showers at the South Pole. The inclusion of the radio technique necessitates an expanded set of simulation and analysis tools to explore the radio-frequency emission from air showers in the 70 MHz to 350 MHz band. In this paper we describe the software modules that have been developed to work with time- and frequency-domain information within IceCube's existing software framework, IceTray, which is used by the entire IceCube collaboration. The software includes a method by which air-shower simulation, generated using CoREAS, can be reused via waveform interpolation, thus overcoming a significant computational hurdle in the field.

Interpolation of Noisy Measurements in a Coarse Grid: Creating Waveforms for Intermediate Points on “747” Curves

In some situations, only a limited number of measurements under a limited number of conditions may be available for analysis. This paper describes an approach for creating additional interpolated measurements in which both measurement noise as well as signal have appropriate values. This is important, for example, when evaluating the behavior of advanced readback channels where quantities, such as sector failure rate (SFR), change very abruptly from 0% to 100% in response to relatively small parameter changes. This means that a smooth interpolation of SFR between coarse measurement points is little help in determining the actual boundary of good performance. If, instead, interpolated waveforms are fed into the channel, the boundary of the region of good performance can be clearly identified. This paper formulates a solution for the general problem of noisy measurement (waveform) interpolation and then applies this approach specifically to the situation, where a series of spin-stand waveforms were captured at various reader positions and with various written track squeeze positions (i.e., a set of “747s”). Using this approach, realistic waveforms can now be created and the SFR evaluated at any arbitrary intermediate reader and writer positions.

A Waveform Interpolation Based Packet Loss Recovery Technique

A Waveform Interpolation Based Packet Loss Recovery Technique

USE OF SINGULAR-VALUE DECOMPOSITION IN GRAVITATIONAL-WAVE DATA ANALYSIS

Singular-value decomposition is a powerful technique that has been used in the analysis of matrices in many fields. In this paper, we summarize how it has been applied to the analysis of gravitational-wave data analysis. These include producing basis waveforms for matched filtering, decreasing the computational cost of searching for many waveforms, improving parameter estimation, and providing a method of waveform interpolation.

Waveform Interpolation-Based Speech Analysis/Synthesis for HMM-Based TTS Systems

This letter proposes an HMM-based Text-to-Speech (TTS) system using waveform interpolation (WI)-based speech analysis and synthesis. The synthesized speech quality of the proposed system is significantly improved due to adopting an enhanced excitation modeling technique. The decomposition of characteristic waveform (CW) into slowly evolving waveform (SEW) and rapidly evolving waveform (REW) is efficient not only for excitation modeling but also for training process of HMMs. Objective and subjective test results verify the superiority of the proposed approach to conventional ones.

Waveform interpolation for applications of time-reversal acoustics.

Acoustic waves can force fluids and small objects along the direction of sound propagation (streaming). Potential applications include forcing the flow of fluid and small particles for drug delivery into the brain, or of underground fluids, such as oil, toward a borehole collection point. Well-focused beams are needed for amplitudes to be large enough to be effective. Time-reversal acoustics (TRA) permits focusing of sound in heterogeneous media, but requires a sensor near a point of desired focus. There is a need, addressed in this talk, to send signals off the grid of available sensors so that forces may be delivered in regions between receivers, and no receiver need be at a point where focus is desired. The Acoustic Shepherd introduces interpolation or data matching of TRA-derived Green functions for this purpose. One particular scheme of geometric-mean interpolation is analyzed, and the curve along which the focus moves from one receiver to another as the interpolating weights are varied is shown. General approximation methods from statistical estimation theory are also described. These methods are expected to work better in heterogeneous media (having unknown acoustic properties) than well-known beam forming methods designed for homogeneous media.

An Enhanced Distortion Measure Based VBR for Waveform Interpolative Speech Coders

In our previous study, a distortion measure based variable bit rate (DM-VBR) scheme in waveform interpolation (WI) coders was proposed. In this paper, the repetition method is proposed to estimate non-transmitted parameters instead of the extrapolation method. For the further reduction of slowly evolving waveform (SEW) bit rates, the dimensions of the past parameters, which are different from those of the current parameters, are converted to match the dimension of the current ones. Distortions between interpolated sub-frames and original sub-frames are measured for the reduction of the SEW parameters. And the usefulness of several other distortion measures is also investigated instead of the simple log spectral distortion. Experimental results show that the coder adopting the new schemes offers above 41% bit rate reduction with almost unnoticeable output speech degradation.

New Variable-Bit-Rate Scheme for Waveform Interpolative Coders

In this paper, we propose a new variable-bit-rate speech coder based on the waveform interpolation concept. After the coder extracts all parameters, the amounts of distortions between the current and the predicted parameters, which are estimated by extrapolation using the past two parameters, are measured for all parameters. A parameter is not transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with the past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while maintaining the speech quality degradation below the 0.1 perceptual evaluation of speech quality (PESQ) score.

Study of a noncontact type micro-CMM with arch-bridge and nanopositioning stages

A high precision micro-CMM (coordinate measuring machine) is under development. The expected measuring range is 25×25×10 mm and the resolution is about 1 nm. For the enhancement of machine accuracy, some new design concepts are introduced. The arch-bridge is proven to have higher stiffness than the conventional rectangular bridge. The proposed co-planar stage can reduce the Abbé error in the vertical direction. The long travel of each axis is activated by the piezo-ceramic ultrasonic motor with its AC drive mode. The fine positioning is achieved by the same motor with its DC drive mode, which can perform nanometer steps proportional to the input voltages. A linear diffraction grating interferometer (LDGI) is developed as the position feedback sensor with the resolution to 1 nm after the waveform interpolation. The autofocusing probe is adopted from DVD pickup head and modified to achieve long-range and short-range detections. Using its focus error signals (FES), the probe can reach to 30 nm accuracy in the short range. In this paper, the design principles of the developed micro-CMM are described.

Low bit-rate voice compression based on frequency domain interpolative techniques

This paper presents an approach, referred to as frequency domain interpolation (FDI), for achieving high-quality speech at low bit-rates (4 kb/s and below) within reasonable complexity and delay. FDI methods, like the prototype waveform interpolation (PWI) methods, derive a prototype waveform (PW) at regular intervals of time. But, unlike PWI, there is no separation into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW) component. Instead, the PW is encoded after gain normalization in magnitude-phase form. The magnitude is modeled as a sum of mean and deviation values in multiple frequency bands and this model is quantized using switched backward adaptive VQ techniques. The phase information is represented as a composite vector of PW correlations in multiple frequency bands and an overall voicing measure. This information is quantized using a VQ at the encoder. At the decoder, a phase model is employed that uses the received phase (and magnitude) information to reproduce PWs with the correct periodicity and evolutionary characteristics. Speech is synthesized by interpolating the reconstructed PWs after gain adjustment and filtering it using the short-term predictor and a postfilter. The design of a 4-kb/s and a 2.4-kb/s FDI codec are presented in this paper and their performance is characterized in terms of delay, complexity, and subjective voice quality. The results confirm that FDI techniques have the potential for delivering high-quality speech at low bit-rates in a cost-effective manner.

Voice Morphing Using 3D Waveform Interpolation Surfaces and Lossless Tube Area Functions

Voice morphing is the process of producing intermediate or hybrid voices between the utterances of two speakers. It can also be defined as the process of gradually transforming the voice of one speaker to that of another. The ability to change the speaker's individual characteristics and to produce high-quality voices can be used in many applications. Examples include multimedia and video entertainment, as well as enrichment of speech databases in text-to-speech systems. In this study we present a new technique which enables production of a given number of intermediate voices or of utterances which gradually change from one voice to another. This technique is based on two components: (1) creation of a 3D prototype waveform interpolation (PWI) surface from the LPC residual signal, to produce an intermediate excitation signal; (2) a representation of the vocal tract by a lossless tube area function, and an interpolation of the parameters of the two speakers. The resulting synthesized signal sounds like a natural voice lying between the two original voices.

Synthesis of speech from pitch prototype waveforms by time-synchronous waveform interpolation

In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes. A pre-selection filter may be applied to determine whether to abandon the TSWI technique in favor of another algorithm for the current frame. A post-selection performance measure may be obtained and compared with a predetermined threshold to determine whether the TSWI algorithm is performing adequately.

Waveform interpolation for 4 kbit/s wideband speech coding

An implementation of waveform interpolation (WI) applied to wideband speech coding at 4 kbit/s is presented. Listening tests comparing narrowband speech coded using standard coders are presented, with wideband speech coded using WI. Results show a clear preference for wideband speech over narrowband speech coded at an equivalent bit rate.

A comparison of spectral smoothing methods for segment concatenation based speech synthesis

There are many scenarios in both speech synthesis and coding in which adjacent time-frames of speech are spectrally discontinuous. This paper addresses the topic of improving concatenative speech synthesis with a limited database by proposing methods to smooth, adjust, or interpolate the spectral transitions between speech segments. The objective is to produce natural-sounding speech via segment concatenation when formants and other spectral features do not align properly. We consider several methods for adjusting the spectra at the boundaries between waveform segments. Techniques examined include optimal coupling, waveform interpolation (WI), linear predictive parameter interpolation, and psychoacoustic closure. Several of these algorithms have been previously developed for either coding or synthesis, while others are enhanced. We also consider the connection between speech science and articulation in determining the type of smoothing appropriate for given phoneme–phoneme transitions. Moreover, this work incorporates the use of a recently-proposed auditory-neural based distance measure (ANBM), which employs a computational model of the auditory system to assess perceived spectral discontinuities. We demonstrate how actual ANBM scores can be used to help determine the need for smoothing. In addition, formal evaluation of four smoothing methods, using the ANBM and extensive listener tests, reveals that smoothing can distinctly improve the quality of speech but when applied inappropriately can also degrade the quality. It is shown that after proper spectral smoothing, or spectral interpolation, the final synthesized speech sounds more natural and has a more continuous spectral structure.

A new approach to generating Pitch Cycle Waveform (PCW) for Waveform Interpolation codec

A novel approach to generate Pitch Cycle Waveform (PCW) for Waveform Interpolation (WI) speech coding is presented in this paper. This approach has the advantage of low delay and low computational complexity. PCW-based synthesis using the proposed method is capable of producing highly intelligible speech at a low bit rate. For better efficiency and perception, voiced and unvoiced sounds are dealt separately and different coding schemes are applied to encode the PCWs at different frequency bands. A 2.0 kbps speech coder has been developed and evaluated by subjective listening tests. The output speech has a fairly good quality that is sufficient for communication purposes.

Concatenative text-to-speech synthesis based on waveform interpolation (a time frequency approach)

The time domain pitch synchronous overlap and add (TD-PSOLA) is the technique most used in comercial concatenative text-to-speech (TTS) synthesis systems. However, it is well known that TD-PSOLA presents several drawbacks. In order to overcome some drawbacks of the TD-PSOLA, this work presents a method based on time frequency interpolation (TFI) [Yair Shoham]. The method introduced here is a pitch-synchronous time-frequency approach of the waveform interpolation technique (WI) [Bastian Kleijn]. The goal of this work is to show that the TFI technique presents some important advantages to concatenative TTS synthesis. It allows pitch scale modification (PSM) independent of time scale modification (TSM) in a quite straightforward manner, and with high quality. TSM and PSM can be done in a continuous way, without any limitation of pitch period resolution. Moreover, the TFI technique allows simple, flexible, and efficient procedures to smooth diphone (or any other kind of unit) boundaries. The proposed system was evaluated using diphones and prosodies generated by the Festival system [Alan Black, Paul Taylor]. Subjective tests were performed, between the proposed TFI system and the standard TD-PSOLA system, highlighting the superior quality of the proposed system in comparison with TD-PSOLA.

On methods for perfect reconstruction WI speech coding with preprocessing

The waveform interpolation (WI) speech coding algorithm has been shown to be an efficient method to describe the evolution of periodic voiced components in the speech signal. However, the conventional WI coding does not provide perfect reconstruction property, i.e. the decoded signal does not converge to the original signal with decreasing quantization error. Therefore errors in the coding model cannot be fixed by quantization. In this paper we discuss about characteristics of the WI coding model and about modifications to the model which enable the perfect reconstruction property. The new requirements and features are examined and discussed in detail. While the perfect reconstruction property brings many benefits it also causes new demands to the operation of the coder. Particularly high requirements are set to the exactness of the pitch estimate; inaccuracies hamper rapidly the possibilities to quantize the parameters efficiently. To overcome this we introduce a preprocessing method which slightly modifies the pitch structure of the residual signal before waveform extraction. The modifications to the signal are minor and therefore the quality of the preprocessed signal is very close to that of the input speech. In the proposed method the perfect reconstruction property is maintained in relation to the preprocessed signal.

Low rate WI SEW representation using a REW-implicit pulse model

Reducing the bit rate of waveform interpolation speech coders while maintaining the perceptual quality has been the focus of a great deal of research. This letter proposes a new method of slowly evolving waveform (SEW) quantization specifically targeted at low rate coding. The proposed method uses a pulse model whose parameters are implicitly contained in the quantized rapidly evolving waveform (REW) parameters, thus requiring no bits for transmission. Results indicate no degradation in perceptual speech quality when compared to that of the existing SEW quantization method. This retention of perceptual quality is in spite of a 12% reduction in the overall coder bit rate.

Accurate, critically sampled characteristic waveformsurface construction forwaveform interpolation decomposition

A waveform-matched waveform interpolation (WMWI) technique is presented which offers improved signal analysis over standard WI coders and, in the unquantised case, perfect reconstruction. In WMWI, an accurate representation of speech evolution is formed by extracting consecutive pitch periods of a time-warped, constant pitch residual. A pitch track optimisation technique is described which ensures that the critically sampled pitch periods can be effectively decomposed into a slowly evolving and rapidly evolving waveform, allowing efficient quantisation.

A new waveform interpolation coding scheme based on pitch synchronous wavelet transform decomposition

This correspondence uses a pitch synchronous wavelet transform (PSWT) as an alternative characteristic waveform decomposition method for the waveform interpolation (WI) paradigm. The proposed method has the benefit of providing additional scalability in quantization than the existing WI decomposition to meet desired quality requirements. The PSWT is implemented as a quadrature mirror filter bank and decomposes the characteristic waveform surface into a series of reduced time resolution surfaces. Efficient quantization of these surfaces is achieved by exploiting their perceptual importance and inherent transmission rate requirements. The multiresolution representation has the additional benefit of more flexible parameter quantization, allowing a more accurate description of perceptually important scales, especially at higher coding rates. The proposed PSWT-WI coder is very well suited to high quality speech storage applications.