Synthetic speech detection and audio steganography in VoIP scenarios

The distinction between synthetic and human voice uses the techniques of the current biometric voice recognition systems, which prevent that a person’s voice, no matter if with good or bad intentions, can be confused with someone else’s. Steganography gives the possibility to hide in a file without a particular value (usually audio, video or image files) a hidden message in such a way as to not rise suspicion to any external observer. This article suggests two methods, applicable in a VoIP hypothetical scenario, which allow us to distinguish a synthetic speech from a human voice, and to insert within the Comfort Noise a text message generated in the pauses of a voice conversation. The first method takes up the studies already carried out for the Modulation Features related to the temporal analysis of the speech signals, while the second one proposes a technique that derives from the Direct Sequence Spread Spectrum, which consists in distributing the signal energy to hide on a wider band transmission. Due to space limits, this paper is only an extended abstract. The full version will contain further details on our research

Band-pass filtering of the time sequences of spectral parameters for robust wireless speech recognition

In this paper we address the problem of automatic speech recognition when wireless speech communication systems are involved. In this context, three main sources of distortion should be considered: acoustic environment, speech coding and transmission errors. Whilst the first one has already received a lot of attention, the last two deserve further investigation in our opinion. We have found out that band-pass filtering of the recognition features improves ASR performance when distortions due to these particular communication systems are present. Furthermore, we have evaluated two alternative configurations at different bit error rates (BER) typical of these channels: band-pass filtering the LP-MFCC parameters or a modification of the RASTA-PLP using a sharper low-pass section perform consistently better than LP-MFCC and RASTA-PLP, respectively.Publicad

Project OASIS: The Design of a Signal Detector for the Search for Extraterrestrial Intelligence

An 8 million channel spectrum analyzer (MCSA) was designed the meet to meet the needs of a SETI program. The MCSA puts out a very large data base at very high rates. The development of a device which follows the MCSA, is presented

Perceptually motivated Sub-band Decomposition for FDLP Audio Coding

This paper describes employment of non-uniform QMF decomposition to increase the efficiency of a generic wide-band audio coding system based on Frequency Domain Linear Prediction (FDLP). The base line FDLP codec, operating at high bit-rates (~136 kbps), exploits a uniform QMF decomposition into 64 sub-bands followed by sub-band processing based on FDLP. Here, we propose a non-uniform QMF decomposition into 32 frequency sub-bands obtained by merging 64 uni- form QMF bands. The merging operation is performed in such a way that bandwidths of the resulting critically sampled sub-bands emulate the characteristics of the critical band filters in the human auditory system. Such frequency decomposition, when employed in the FDLP audio codec, results in a bit-rate reduction of 40% over the base line. We also describe the complete audio codec, which provides high-fidelity audio compression at ~66 kbps. In subjective listening tests, the FDLP codec outperforms MPEG-1 Layer 3 (MP3) and achieves similar qualities as MPEG-4 HE-AAC codec

Non-uniform QMF Decomposition for Wide-band Audio Coding based on Frequency Domain Linear Prediction

This paper presents a new technique for perfect reconstruction non-uniform QMF decomposition developed to increase efficiency of a generic wide-band audio coding system based on Frequency Domain Linear Prediction (FDLP). The base line FDLP codec, operating at high bit-rates (~136 kbps), exploits an uniform QMF decomposition into 64 sub-bands followed by sub-band processing based on FDLP. Here, we propose a non-uniform QMF decomposition into 32 frequency sub-bands obtained by merging 64 uniform QMF bands. The merging operation is performed in such a way that bandwidths of the resulting critically sampled sub-bands emulate the characteristics of the critical band filters in the human auditory system. Such frequency decomposition, when employed in the FDLP audio codec, results in a bit-rate reduction of 40% over the base line. We also describe the complete audio codec, which provides high-fidelity audio compression at ~66 kbps. In subjective listening tests, the FDLP codec outperforms MPEG-1 Layer 3 (MP3) and achieves similar qualities as MPEG-4 AAC+ standard

Electroacoustic Assessment of Hearing Aids and PSAPs

Hearing aids and personal sound amplification products (PSAPs) are commonly used assistive devices for treating hearing loss. Due to the diversity in the hardware and signal processing algorithms in these devices, comprehensive verification of their performance is essential. Existing standards for assistive hearing devices are primarily used for quality control purposes and do not quantify their performance in a perceptually-relevant manner. This thesis developed a comprehensive electroacoustic testing toolbox for hearing devices that encompasses both quality control and perceptually-relevant measures. In particular, a test sequence was developed to assess the effectiveness of noise reduction feature in assistive hearing devices. Several commercially-available hearing aids and PSAPs on the “best seller” list at Amazon.ca were evaluated using the toolbox. Key results include: (a) hearing aids differ in their noise reduction performance; (b) some of the popular PSAPs do not meet the ANSI standards and are capable of producing dangerous sound pressure levels; and (c) hearing aids performed better than PSAPs on perceptually-relevant metrics

Some Commonly Used Speech Feature Extraction Algorithms

Speech is a complex naturally acquired human motor ability. It is characterized in adults with the production of about 14 different sounds per second via the harmonized actions of roughly 100 muscles. Speaker recognition is the capability of a software or hardware to receive speech signal, identify the speaker present in the speech signal and recognize the speaker afterwards. Feature extraction is accomplished by changing the speech waveform to a form of parametric representation at a relatively minimized data rate for subsequent processing and analysis. Therefore, acceptable classification is derived from excellent and quality features. Mel Frequency Cepstral Coefficients (MFCC), Linear Prediction Coefficients (LPC), Linear Prediction Cepstral Coefficients (LPCC), Line Spectral Frequencies (LSF), Discrete Wavelet Transform (DWT) and Perceptual Linear Prediction (PLP) are the speech feature extraction techniques that were discussed in these chapter. These methods have been tested in a wide variety of applications, giving them high level of reliability and acceptability. Researchers have made several modifications to the above discussed techniques to make them less susceptible to noise, more robust and consume less time. In conclusion, none of the methods is superior to the other, the area of application would determine which method to select

Étude de transformées temps-fréquence pour le codage audio faible retard en haute qualité

In recent years there has been a phenomenal increase in the number of products and applications which make use of audio coding formats. Amongthe most successful audio coding schemes, the MPEG-1 Layer III (mp3), the MPEG-2 Advanced Audio Coding (AAC) or its evolution MPEG-4High Efficiency-Advanced Audio Coding (HE-AAC) can be cited. More recently, perceptual audio coding has been adapted to achieve codingat low-delay such to become suitable for conversational applications. Traditionally, the use of filter bank such as the Modified Discrete CosineTransform (MDCT) is a central component of perceptual audio coding and its adaptation to low delay audio coding has become an important researchtopic. Low delay transforms have been developed in order to retain the performance of standard audio coding while reducing dramatically the associated algorithmic delay.This work presents some elements allowing to better accommodate the delay reduction constraint. Among the contributions, a low delay blockswitching tool which allows the direct transition between long transform and short transform without the insertion of transition window. The sameprinciple has been extended to define new perfect reconstruction conditions for the MDCT with relaxed constraints compared to the original definition.As a consequence, a seamless reconstruction method has been derived to increase the flexibility of transform coding schemes with the possibility toselect a transform for a frame independently from its neighbouring frames. Finally, based on this new approach, a new low delay window design procedure has been derived to obtain an analytic definition for a new family of transforms, permitting high quality with a substantial coding delay reduction. The performance of the proposed transforms has been thoroughly evaluated, an evaluation framework involving an objective measurement of the optimal transform sequence is proposed. It confirms the relevance of the proposed transforms used for audio coding. In addition, the new approaches have been successfully applied to the recent standardisation work items, such as the low delay audio coding developed at MPEG (LD-AAC and ELD-AAC) and they have been evaluated with numerous subjective testing, showing a significant improvement of the quality for transient signals. The new low delay window design has been adopted in G.718, a scalable speech and audio codec standardized in ITU-T and has demonstrated its benefit in terms of delay reduction while maintaining the audio quality of a traditional MDCT.Codage audio à faible retard à l'aide de la définition de nouvelles fenêtres pour la transformée MDCT et l'introduction d'un nouveau schéma de commutation de fenêtre