Audio Time-Scale Modification

Audio time-scale modification is an audio effect that alters the duration of an audio signal without affecting its perceived local pitch and timbral characteristics. There are two broad categories of time-scale modifications algorithms, time-domain and frequency-domain. The computationally efficient time-domain techniques produce high quality results for single pitched signals such as speech, but do not cope well with more complex signals such as polyphonic music. The less efficient frequency-domain techniques have proven to be more robust and produce high quality results for a variety of signals; however they introduce a reverberant artefact into the output. This dissertation focuses on incorporating aspects of time-domain techniques into frequency-domain techniques in an attempt to reduce the presence of the reverberant artefact and improve upon computational demands. From a review of prior work it was found that there are a number of time-domain algorithms available and that the choice of algorithm parameters varies considerably in the literature. This finding prompted an investigation into the effects of the choice of parameters and a comparison of the various techniques employed in terms of computational requirements and output quality. The investigation resulted in the derivation of an efficient and flexible parameter set for use within time-domain implementations. Of the available frequency-domain approaches the phase vocoder and time-domain/subband techniques offer an efficiency and robustness advantage over sinusoidal modelling and iterative phase update techniques, and as such were identified as suitable candidates for the provision of a framework for further investigations. Following from this observation, improvements in the quality produced by time-domain/subband techniques are realised through the use of a bark based subband partitioning approach and effective subband synchronisation techniques. In addition, computational and output quality improvements with a phase vocoder implementation are achieved by taking advantage of a certain level of flexibility in the choice of phase within such an implementation. The phase flexibility established is used to push or pull phase values into a phase coherent stage. Further improvements are realised by incorporating features of time-domain algorithms into the system in order to provide a ‘good’ initial set of phase estimates; the transition to ‘perfect’ phase coherence is significantly reduced through this scheme, thereby improving the overall output quality produced. The result is a robust and efficient time-scale modification algorithm which draws upon various aspects of a number of general approaches to time-scale modification

Time-scale and pitch modifications of speech signals and resynthesis from the discrete short-time Fourier transform

The modification methods described in this paper combine characteristics of PSOLA-based methods and algorithms that resynthesize speech from its short-time Fourier magnitude only. The starting point is a short-time Fourier representation of the signal. In the case of duration modification, portions, in voiced speech corresponding to pitch periods, are removed from or inserted in this representation. In the case of pitch modification, pitch periods are shortened or extended in this representation, and a number of pitch periods is inserted or removed, respectively. Since it is an important tool for both duration and pitch modification, the resynthesis-from-short-time-Fourier-magnitude-only method of Griffin and Lim (1984) and Griffin et al. (1984) is reviewed and adapted. Duration and pitch modification methods and their results are presented.\ud \u

A Phase Vocoder based on Nonstationary Gabor Frames

We propose a new algorithm for time stretching music signals based on the theory of nonstationary Gabor frames (NSGFs). The algorithm extends the techniques of the classical phase vocoder (PV) by incorporating adaptive time-frequency (TF) representations and adaptive phase locking. The adaptive TF representations imply good time resolution for the onsets of attack transients and good frequency resolution for the sinusoidal components. We estimate the phase values only at peak channels and the remaining phases are then locked to the values of the peaks in an adaptive manner. During attack transients we keep the stretch factor equal to one and we propose a new strategy for determining which channels are relevant for reinitializing the corresponding phase values. In contrast to previously published algorithms we use a non-uniform NSGF to obtain a low redundancy of the corresponding TF representation. We show that with just three times as many TF coefficients as signal samples, artifacts such as phasiness and transient smearing can be greatly reduced compared to the classical PV. The proposed algorithm is tested on both synthetic and real world signals and compared with state of the art algorithms in a reproducible manner.Comment: 10 pages, 6 figure

Application of time-scale and frequency-scale modification alogrithms to voice-gender conversion

Voice conversion is an active new branch of speech processing and deals with the transformation of natural speech, focusing on changing the characteristics of the speaker’s voice. As a result of voice conversion, the speaker’s identity can be changed to make the converted speech sound as if it were uttered by a different speaker, or certain characteristics of the voice can be modified while maintaining the speaker’s identity. Voice-gender conversion (VGC) is a subset of voice conversion and focuses on the transformation of gender-specific voice characteristics. As a result of a voice-gender conversation, male speech is converted into female-sounding speech and vice versa. A major application of voice conversion is speaker normalisation. This means that a given voice is converted to a normalised voice. This allows speech recognition and speech compression methods to perform better as their effective signal space is reduced significantly. In speech compression applications, the reduction of the signal space enhances the efficiency and achieves higher compression rates. Another application is voice transformation to accommodate hearing impairements: a straightforward application is the usage of voice-gender conversion to disguise voices for the protection of individuals, e. g. witnesses, or for nuisance-call determent. The system presented in this thesis achieves voice-gender transformation by independently frequency-scaling the excitation and the formant spectrum of the speech signal in order to model the different voice-gender features from the voice-production perspective. The novelty of this research is the linearization of the non-linear relationship between the male and female formant spectrum. The algorithm used to achieve frequency-scaling is a time scale modification (TSM) algorithm called adaptive over-lap and add (AOLA), which is a recently developed method to efficiently change the duration of time-based signals

Audio Analysis/synthesis System

A method and apparatus for the automatic analysis, synthesis and modification of audio signals, based on an overlap-add sinusoidal model, is disclosed. Automatic analysis of amplitude, frequency and phase parameters of the model is achieved using an analysis-by-synthesis procedure which incorporates successive approximation, yielding synthetic waveforms which are very good approximations to the original waveforms and are perceptually identical to the original sounds. A generalized overlap-add sinusoidal model is introduced which can modify audio signals without objectionable artifacts. In addition, a new approach to pitch-scale modification allows for the use of arbitrary spectral envelope estimates and addresses the problems of high-frequency loss and noise amplification encountered with prior art methods. The overlap-add synthesis method provides the ability to synthesize sounds with computational efficiency rivaling that of synthesis using the discrete short-time Fourier transform (DSTFT) while eliminating the modification artifacts associated with that method.Georgia Tech Research Corporatio

Tatouage audio par EMD

In this paper a new adaptive audio watermarking algorithm based on Empirical Mode Decomposition (EMD) is introduced. The audio signal is divided into frames and each one is decomposed adaptively, by EMD, into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs). The watermark and the synchronization codes are embedded into the extrema of the last IMF, a low frequency mode stable under different attacks and preserving audio perceptual quality of the host signal. The data embedding rate of the proposed algorithm is 46.9–50.3 b/s. Relying on exhaustive simulations, we show the robustness of the hidden watermark for additive noise, MP3 compression, re-quantization, filtering, cropping and resampling. The comparison analysis shows that our method has better performance than watermarking schemes reported recently