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Since speaker recognition and verification became heavily used technology, both in professional applications like forensics and more everyday ones, the question arose: what factors can impact results of those processes? One thing that may be important with respect to this subject is lossy coding, as some of the information contained in an original file is lost in the coding process. In the era of globalization, not only native languages or languages of neighboring countries are of interest to researchers, but also those quite far, especially from Asia - the biggest exporter of goods and services to Europe. Those economic relationships are usually connected with the interchange of personnel, which further shortens geographical distance. The article presents the results that are a continuation of research on the behavior of Japanese language formants. Earlier research focused on changes occurring for the first and second formants. This article presents changes observed for the third and fourth formants. The knowledge of these changes is indicated in the process of speaker identification in forensics using the spectrographic method. At the Department of Acoustics and Multimedia, Wroclaw University of Science and Technology and in many centers around the world, the auditoryspectrographic method is used, which is a combination of the aural and spectrographic methods. In the spectrographic part, a person is identified on the basis of a comparison of the formants' trajectory. [ABSTRACT FROM AUTHOR]

This paper introduces a method for real-time speech coding that combines a binary-latent-vector variational recurrent neural network for mel-spectrogram coding with a non-autoregressive convolutional vocoder for waveform reconstruction. To enable bitrate scalability, we propose a latent vector truncation and padding technique. We evaluate both fixed- and scalable-bitrate variants of the proposed method, comparing them to a baseline vector quantization-based coder. The method is also benchmarked against Opus, Lyra v2, EnCodec, and AudioDec using objective metrics and subjective ratings from a MUSHRA listening test. At 1.38 kbps, the proposed method significantly outperforms Lyra v2 at 3kbps and at 5.51kbps matches its performance at 6kbps. Although AudioDec significantly surpasses the proposed method at 6.4kbps on test data from the TSP speech dataset, the proposed method shows competitive or superior results on withheld speakers from the VCTK dataset. The results show that recurrent coding with binary latent vectors is a viable alternative to prevailing vector quantization-based approaches.

Recently, a conventional neural decoder for speech codec has shown promising performance. However, it typically requires some prior knowledge of decoding such as bit allocation or dequantization scheme, which is not a universal solution for many different kinds of speech codecs. In order to address this limitation, we propose a neurally optimized decoder based on a generative model which can directly reconstruct the speech from the bitstream without a prior knowledge. The proposed decoder mainly consists of two components: 1) a dequantization model to group and dequantize related bits from the bitstream and 2) a generative model to restore the speech conditioned on the output of the dequantization model. Through experiments with mixed excitation linear prediction (MELP), Advanced multi-band excitation (AMBE), and SPEEX at around 2.4 kb/s, it is showed that the proposed model showed better performance in most of the objective and subjective evaluation compared to the conventional speech codecs. [ABSTRACT FROM AUTHOR]

In order to protect the data security and privacy of Voice over Internet Protocol (VoIP), a new algorithm for speech information hiding is proposed based on complete binary tree dynamic codebook grouping (CBTDCG). The algorithm uses the quantization index modulation (QIM) method to divide the dynamic codebook into two parts (corresponding to the embedded secret information bits `0' and `1' respectively), and adopts the complete binary tree to achieve the optimal grouping. The utility of complete binary tree makes the algorithm have the characteristics of fast search and high accuracy. The sender selects the corresponding code word according to the secret information bits for embedding the secret information into carrier speech. At the receiving end, the secret information bits are extracted according to the features of the code word vectors by the receiver. The experimental results show that the hidden capacity of the algorithm is larger than 450bps and the loss of the PESQ is less than 4%. Verification shows that the algorithm possesses a good robustness with a low bit error rate after being interfered by noise and the speech stream carrying secret information has better imperceptibility.

This paper proposes a convolutional neural network (CNN)-based encoder model to compress and code speech signal directly from raw input speech. Although the model can synthesize wideband speech by implicit bandwidth extension, narrowband is preferred for IP telephony and telecommunications purposes. The model takes time domain speech samples as inputs and encodes them using a cascade of convolutional filters in multiple layers, where pooling is applied after some layers to downsample the encoded speech by half. The final bottleneck layer of the CNN encoder provides an abstract and compact representation of the speech signal. In this paper, it is demonstrated that this compact representation is sufficient to reconstruct the original speech signal in high quality using the CNN decoder. This paper also discusses the theoretical background of why and how CNN may be used for end-to-end speech compression and coding. The complexity, delay, memory requirements, and bit rate versus quality are discussed in the experimental results.

With nearly two and a half billion users--a third of the world's population--Facebook far and away hosts the largest speech platform in the history of humanity. In the United States, seven out of ten adults use it, and nearly half get news from it. It is therefore no exaggeration to observe that Facebook's selfpromulgated rules for "what is and is not allowed on Facebook," its contentbased Community Standards, rival if not exceed the First Amendment's importance in shaping discourse in the United States. Yet, unlike the First Amendment, which only protects against government censorship, no scholarship has scrutinized the Community Standards' private regulation of speech on Facebook's far-reaching social media platform. As a result, basic questions about Facebook's sprawling speech code remain unexamined. For instance, at a minimum, do they offer enough clarity to ordinary users and prevent arbitrary or discriminatory moderation? Are they coherent and consistent as a whole? Furthermore, what do they reveal about the nature of Facebook's professed "commitment to expression"? And how do the Community Standards compare--for better or for worse--with modern First Amendment principles?. This Article inaugurates scholarly consideration of these basic questions about the Community Standards in the critical context of the spread of deliberate falsehoods and other disinformation on the platform to influence the electorate. Spoiler alert: Facebook's speech code suffers from a basic lack of clarity and do not embody a coherent or consistent commitment to expression. Indeed, in many respects, the Community Standards suppress significantly more speech--including a wide range of political speech--than the First Amendment would permit of government censors. In addition, Facebook categorically exempts "politicians" from its Community Standards as well as its fact-checking policy out of "respect for the democratic process." But this selective hands-off approach perversely skews public debate by amplifying the expressive power of already dominant speakers in our society. Politicians enjoy an unrestricted license to exploit Facebook's vast reach and highly effective ad targeting tools to spread expedient falsehoods among the most receptive users. Making matters worse, Facebook by design does not broaden or balance public discussion but maximizes user engagement through algorithmically feeding users content tailored to their interests and likes. Many users thereby end up in politically imbalanced, self-reinforcing content bubbles where sensational falsehoods can gain credibility and velocity. While Facebook's special solicitude for politicians and polarizing personalized feed might advance the company's political and business interests, they also promote a race to the bottom among willing politicians and further divide the electorate into balkanized realities based on "alternative facts.". Ultimately, Facebook's speech code and policies for its social media platform present serious shortcomings, particularly in the electoral context. There is much room--and need--for further scrutiny. [ABSTRACT FROM AUTHOR]

Highlights • A packet-loss robust scalable wideband speech codec is proposed. • Performance is improved using the wavelet transform instead of the MDCT. • The proposed codec outperforms the state-of-the-art codec in objective tests. • High performance of the proposed codec is confirmed in subjective tests. Abstract Most recent speech codecs employ code excited linear prediction (CELP) and transmit side information to improve speech quality under packet loss. Another approach to achieve high robustness to packet loss is to use the frame independent coding scheme based on the internet low bitrate codec (iLBC). The scalable wideband speech codec based on the iLBC was previously presented and outperformed G.729.1 at most bit rates according to the objective quality. This paper presents improvements to the previous work. Specifically, we employ the wavelet packet transform (WPT) instead of the modified discrete cosine transform (MDCT) to enhance the quality, and evaluate the proposed codec based on both the objective and subjective quality measures. The objective quality evaluation results show that clear improvement is achieved and that the proposed codec outperforms G.729.1 at the bit rate of 18 kbps or higher under clean channel conditions and has higher robustness to packet loss than G.729.1. The informal subjective test results also show similar trends. [ABSTRACT FROM AUTHOR]

In this paper, we examine the impact of four voice over long term evolution adaptive multi-rate wideband codec mode-sets on coverage at pedestrian and vehicular speeds. Industry-standardized mean opinion scores were used as a metric for voice quality. Controlled laboratory experiments simulating pedestrian speeds indicated that there was an improvement in voice quality when mode-set eight was employed. At vehicular speeds, mode-set eight outperformed the other mode-sets for path losses less than 130 dB; however, all four mode-sets experienced a significant decline in voice quality when the path loss was greater than 130 dB. Based on the current implementations, there are no significant benefits to lowering the mode-sets or deploying dynamic codec rate adaptation.

The automatic identification of person's identity from their voice is a part of modern telecommunication services. In order to execute the identification task, speech signal has to be transmitted to a remote server. So a performance of the recognition/identification system can be influenced by various distortions that occur when transmitting speech signal through a communication channel. This paper studies an effect of telecommunication channel, particularly commonly used narrowband (NB) speech codecs in current telecommunication networks, on a performance of automatic speaker recognition in the context of a channel/codec mismatch between enrollment and test utterances. An influence of speech coding on speaker identification is assessed by using the reference GMM-UBM method. The results show that the partially mismatched scenario offers better results than the fully matched scenario when speaker recognition is done on speech utterances degraded by the different NB codecs. Moreover, deploying EVS and G.711 codecs in a training process of the recognition system provides the best success rate in the fully mismatched scenario. It should be noted here that the both EVS and G.711codecs offer the best speech quality among the codecs deployed in this study. This finding also fully corresponds with the finding presented by Janicki & Staroszczyk in [1] focusing on other speech codecs.

Abstract Estimation of glottal source information can be performed non-invasively from speech by using glottal inverse filtering (GIF) methods. However, the existing GIF methods are sensitive even to slight distortions in speech signals under different realistic scenarios, for example, in coded telephone speech. Therefore, there is a need for robust GIF methods which could accurately estimate glottal flows from coded telephone speech. To address the issue of robust GIF, this paper proposes a new deep neural net-based glottal inverse filtering (DNN-GIF) method for estimation of glottal source from coded telephone speech. The proposed DNN-GIF method utilizes both coded and clean versions of speech signal during training. DNN is used to map the speech features extracted from coded speech with the glottal flows estimated from the corresponding clean speech. The glottal flows are estimated from the clean speech by using quasi closed phase analysis (QCP). To generate coded telephone speech, adaptive multi-rate (AMR) codec is utilized which operates in two transmission bandwidths: narrow band (300 Hz - 3.4 kHz) and wide band (50 Hz - 7 kHz). The glottal source parameters were computed from the proposed and existing GIF methods by using vowels obtained from natural speech data as well as from artificial speech production models. The errors in glottal source parameters indicate that the proposed DNN-GIF method has considerably improved the glottal flow estimation under coded condition for both low- and high-pitched vowels. The proposed DNN-GIF method can be utilized to accurately 1 1 In this article, the term "accurate/accuracy" is used only when referring to quantitative, objective measures. extract glottal source -based features from coded telephone speech which can be used to improve the performance of speech technology applications such as speaker recognition, emotion recognition and telemonitoring of neurodegerenerative diseases. [ABSTRACT FROM AUTHOR]

Voice over IP (VoIP) communications are prone to transmission delays and data losses as they are carried out over packet-switched networks which are unable to guarantee real-time packet delivery. Speech codecs used in these channels strongly rely on Packet Loss Concealment (PLC) algorithms, the performance of which can be compromised as frame losses often occur in bursts. Thus, advanced PLC algorithms for erasure channels have already been proposed in the literature but these frequently focus on the speech envelope disregarding the excitation signal. In this paper we propose an error mitigation scheme focused on the estimation of this excitation signal whenever lost frames appear. These estimates are obtained by applying a minimum mean square error (MMSE) estimation technique based on the last correctly received frame. To this end an excitation signal’s representation and quantization approach which compares the resulting synthesized signal with the original speech one is considered. In addition, we propose the combination of this approach with a recursive least squares (RLS) technique which provides a better excitation signal estimate for the first lost consecutive frames. The proposed error mitigation scheme has been tested on the iLBC codec, where objective and subjective tests have shown a noticeable improvement on speech quality for transmissions over erasure channels. [ABSTRACT FROM AUTHOR]

Steganography is a means of covert communication without revealing the occurrence and the real purpose of communication. The adaptive multirate wideband (AMR-WB) is a widely adapted format in mobile handsets and is also the recommended speech codec for VoLTE. In this paper, a novel AMR-WB speech steganography is proposed based on diameter-neighbor codebook partition algorithm. Different embedding capacity may be achieved by adjusting the iterative parameters during codebook division. The experimental results prove that the presented AMR-WB steganography may provide higher and flexible embedding capacity without inducing perceptible distortion compared with the state-of-the-art methods. With 48 iterations of cluster merging, twice the embedding capacity of complementary-neighbor-vertices-based embedding method may be obtained with a decrease of only around 2% in speech quality and much the same undetectability. Moreover, both the quality of stego speech and the security regarding statistical steganalysis are better than the recent speech steganography based on neighbor-index-division codebook partition. [ABSTRACT FROM AUTHOR]

In this study, two novel methods, which are based on Karhunen Loeve Transform (KLT) and Independent Component Analysis (ICA), are proposed for coding of speech signals. Instead of immediately dealing with eigenvalue magnitudes, the KLT- and ICA-based methods use eigenvectors of covariance matrices (or independent components for ICA) by geometrically grouping these vectors into fewer numbers of vectors. In this way, a data representation compaction is achieved. Further compression is achieved through discarding autocovariance eigenvectors corresponding to the small eigenvalues and applying vector quantization on the remaining eigenvectors. Additionally, this study proposes an iterative error refinement process, which uses the rest of the available bandwidth in order to transmit an efficient representation of the description error for better SNR. The overall process constitutes a new approach to efficient speech coding, with ICA being used in subspace speech coding for the first time. Constant bit rate (CBR) and variable bit rate (VBR) coding algorithms are employed with the proposed methods. TIMIT speech database is used in the experimental studies. Speech signals are synthesized at 2.4 kbps, 8 kbps, 12.2 kbps, 16 kbps, 16.4kbps and 19.85 kbps rates by using various frame lengths. The qualities of synthesized speech signals are compared to those of available speech codecs, i.e., LPC (2.4 kbps), G.728 (LD-CELP, 16 kbps), G.729A (CS-CELP, 8 kbps), EVS (16.4 kbps), AMR-NB (12.2 kbps) and AMR-WB (19.85 kbps). [ABSTRACT FROM AUTHOR]

This article puts forward a new algorithm for voice conversion which not only removes the necessity of parallel corpus in the training phase but also resolves the issue of insufficiency of the target speaker's corpus. The proposed approach is based on one of the new voice conversion models utilizing classical LPC analysis-synthesis model combined with GMM. Through this algorithm, the conversion functions among vowels and demi-syllables are derived. We assumed that these functions are rather the same for different speakers if their genders, accents, and languages are alike. Therefore, we will be able to produce the demi-syllables with just having access to few sentences from the target speaker and forming the GMM for one of his/her vowels. The results from the appraisal of the proposed method for voice conversion clarifies that this method has the ability to efficiently realize the speech features of the target speaker. It can also provide results comparable to the ones obtained through the parallel-corpus-based approaches. [ABSTRACT FROM AUTHOR]

This paper proposes modification in the transmission of excitation codevector and its non-zero pulse sign magnitude using 'codebook partition and label assignment' approach, which in turn reduces the number of bits required to transmit it through the communication channel in legacy CS-ACELP 8 kbps speech codec. The proposed approach uses the excitation codebook structure of forward mode standard G.729E 11.8 kbps with two non-zero pulses per track which avoids the use of two algebraic codebook structure for forward mode as well as for backward mode of G.729E with least significant pulse replacement approach for finding optimized excitation codevector. Proposed modification in legacy 8 kbps CS-ACELP (80 bits/10 ms) speech codec actuates the bit rate of 10.6 kbps (106 bits/10 ms) with a better objective and subjective analysis in stark contrast with legacy 8 kbps CS-ACELP speech coder and also avoids the switching of codebook modes of standard 11.8 kbps (G.729E) CS-ACELP speech coder. This paper also aims to propose the reduction in the number of searches in the final codevector of excitation structure by considering initial codevector as a final codevector which improves the quality of the speech compared to the output speech quality of legacy G.729 CS-ACELP working at 8 kbps. Both legacy CS-ACELP 8 kbps speech codec and proposed CS-ACELP 10.6 kbps are implemented in MATLAB. Subjective and objective analysis are carried out on a proposed CS-ACELP 10.6 kbps speech codec in order to evaluate its performance and the results obtained are then cross- compared with the results of legacy CS-ACELP (8 kbps) using set of tables and graphs. It is evident from obtained results that both PESQ and MOS scores are quite comparable for each set of wave files even though bitrates are reduced. Consistency and efficiency of proposed algorithm is assured by calculating the population mean of 95% confidence interval based on obtained objective and subjective parameter results. [ABSTRACT FROM AUTHOR]

Substantial progress has been achieved in voice‐based biometrics in recent times but a variety of challenges still remain for speech research community. One such obstacle is reliable speaker authentication from speech signals degraded by lossy compression. Compression is commonplace in modern telecommunications, such as mobile telephony, VoIP services, teleconference, voice messaging or gaming. In this study, the authors investigate the effect of lossy speech compression on text‐independent speaker verification. Voice biometrics performance is evaluated on clean speech signals distorted by the state‐of‐the‐art narrowband (NB) as well as wideband (WB) speech codecs. The tests are performed in both channel‐matched and channel‐mismatched scenarios. The test results show that coded WB speech improves voice authentication precision by 1–3% of equal error rate over coded NB speech, even at the lowest investigated bitrates. It is also shown that the enhanced voice services codec does not provide better results than the other codecs involved in this study. [ABSTRACT FROM AUTHOR]

In this paper, we focus on quantization-index-modulation (QIM) steganography in low-bit-rate speech codec and contribute to improve its steganalysis resistance. A novel QIM steganography is proposed based on the replacement of quantization index set in linear predictive coding (LPC). In this method, each quantization index set is seen as a point in quantization index space. Steganography is conducted in such space. Comparing with other methods, our algorithm significantly improves the embedding efficiency. One quantization index needs to be changed at most when three binary bits are hidden. The number of alterations introduced by the proposed approach is much lower than that of the current methods with the same embedding rate. Due to the fewer cover changes, the proposed steganography is less detectable. Moreover, a division strategy based on the genetic algorithm is proposed to reduce the additional distortion introduced by replacements. In our experiment, ITU-T G.723.1 is selected as the codec, and the experimental results show that the proposed approach outperforms the state-of-the-art LPC-based approach in low-bit-rate speech codec with respect to both steganographic capacity and steganalysis resistance. [ABSTRACT FROM AUTHOR]

Unified speech and audio coding (USAC) is one of the latest coding technologies. It is based on a switchable coding structure, and has demonstrated the highest levels of performance for both speech and music contents. In this paper, we propose an extended version of USAC with a single-mode of operation-which does not require a switching system-by extending the linear predictioncoding mode. The main concept of this extension is the adoption of the advantages of frequency-domain coding schemes, such as windowing and transition control. Subjective test results indicate that the proposed scheme covers speech, music, and mixed streams with adequate levels of performance. The obtained quality levels are comparable with those of USAC. [ABSTRACT FROM AUTHOR]

An adaptive multi-rate wideband (AMR-WB) speech codec with a sampling rate of 16 kHz is known as one of the speech codecs employed in handheld devices that support 4G mobile communication systems. When applied to smartphones, it provides a superior speech quality relative to conventional speech codecs. Nonetheless, a major disadvantage is that an algebraic codebook search occupies a significant computational load in an AMR-WB encoder. In other words, the high computational complexity accounts for the high power consumption on a smartphone battery. This paper presents an improved version of depth-first tree search (DF) algorithm as a means to considerably reduce the complexity of an algebraic codebook search in an AMR-WB speech codec. This proposed search algorithm firstly involves the choice of a specified number of candidate pulses according to a pulse contribution ranking. Subsequently, a DF search is performed on the candidate pulses for a set of best pulses. Consequently, the target of the search and computational complexity reduction can be reached as expected. With a well maintained speech quality, this proposal demonstrates a search performance superiority over a DF and a global pulse replacement approach. Furthermore, with DF as a benchmark, a computational load reduction above 73% is reached in all coding modes. [ABSTRACT FROM AUTHOR]

For transmission of packetised media, real‐time communications rely on the real‐time protocol (RTP). RTP traffic, however, is transmitted over datagram‐based transport that allows fast delivery of media. Internet firewalls are typically configured to allow web friendly traffic that uses stream‐based transport instead and, simultaneously as security measure, reject any other type of transport. A mechanism to overcome this limitation is by switching transport such that media frames that would be rejected by firewalls are transmitted on top of stream transport. In this study, a novel mathematical model that links application layer packet loss to bursty network packet loss in the context of fading channels characteristic of wireless communications is presented for both, datagram and stream transport. This model is compared with experimental scenarios applied to state‐of‐the‐art speech codecs in order to obtain quality scores that can be correlated against theoretical loss probabilities. [ABSTRACT FROM AUTHOR]

The long-term harmonic plus noise model (LT-HNM) for speech shows an interesting data compression, since it exploits the smooth evolution of the time trajectories of the short-term harmonic plus noise model parameters, by applying a discrete cosine model (DCM). In this paper, we extend the LT-HNM to a complete low bit-rate speech coder. A Normalized Split Vector Quantization (NSVQ) is proposed to quantize the variable dimension LT-DCM vectors. The NSVQ is designed according to the properties of the DCM vectors obtained from a standard speech database. The obtained LT-HNM coder reaches an average bit-rate of 2.7 kbps for wideband speech. The proposed coder is evaluated in terms of modeling and coding errors, bit-rate, listening quality and intelligibility. [ABSTRACT FROM AUTHOR]

An adaptive multi-rate wideband (AMR-WB) code is a speech codec developed on the basis of an algebraic code-excited linear-prediction (ACELP) coding technique, and has a double advantage of low bit rates and high speech quality. This coding technique is widely used in modern mobile communication systems for a high speech quality in handheld devices. However, a major disadvantage is that a vector quantization (VQ) of immittance spectral frequency (ISF) coefficients occupies a significant computational load in the AMR-WB encoder. Hence, this paper presents a triangular inequality elimination (TIE) algorithm combined with a dynamic mechanism and an intersection mechanism, abbreviated as the DI-TIE algorithm, to remarkably improve the complexity of ISF coefficient quantization in the AMR-WB speech codec. Both mechanisms are designed in a way that recursively enhances the performance of the TIE algorithm. At the end of this work, this proposal is experimentally validated as a superior search algorithm relative to a conventional TIE, a multiple TIE (MTIE), and an equal-average equal-variance equal-norm nearest neighbor search (EEENNS) approach. With a full search algorithm as a benchmark for search load comparison, this work provides a search load reduction above 77%, a figure far beyond 36% in the TIE, 49% in the MTIE, and 68% in the EEENNS approach. [ABSTRACT FROM AUTHOR]

This paper presents an improved version of a steganographic algorithm for IP telephony called HideF0. It is based on approximating the F0 parameter, which is responsible for conveying information about the pitch of the speech signal. The bits saved due to simplification of the pitch contour are used for the hidden transmission. In our experiments, the proposed method was applied to the narrowband Speex codec working in five different modes, with bitrates between 5,950 bps and 24,600 bps. We showed that HideF0 was able to create hidden channels with steganographic bandwidths of around 200 bps at the expense of a steganographic cost of between 0.5 and 0.7 MOS, depending on the Speex mode. Because of placing the approximation flag in the voice packet header, the improved version of the proposed algorithm yielded a significantly lower decrease in speech quality, when compared with the original version of HideF0. In addition, for low bitrates of the hidden channel (i.e., below ca. 50 bps) it was able to operate without introducing any steganographic cost. Copyright © 2016 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The rapid development of speech communication technology has made it possible for low bit-rate speech to become appropriate steganographic cover media. To incorporate data hiding into the low bit-rate speech codec, a novel steganography algorithm is proposed in this paper. By analyzing the encoding rule of fixed codebook vector, the way of transposing encoding locations of adjacent pulses is found to be suitable for data embedding with good imperceptibility. Based on encoding location transposition of adjacent pulses, the relationship between adjacent pulse locations is used to embed secret data while the fixed codebook search is being conducted during the encoding process of G.729 codec, which can maintain synchronization between data embedding and speech encoding. The experimental results demonstrate that the proposed steganography algorithm performs well in imperceptibility with a hiding capacity of 550 bits/s. Furthermore, the real-time and anti-detection performances are also satisfactory. [ABSTRACT FROM AUTHOR]

This study proposes a new objective speech quality measure using the responses of a physiologically-based computational model of auditory nerve (AN). The population response of the model AN fibers to a speech signal is represented by a 2D neurogram, and features of the neurogram are extracted by orthogonal moments. A special type of orthogonal moment, the orthogonal Tchebichef-Krawtchouk moment, is used in this study. The proposed measure is compared to the subjective scores from two standard databases, the NOIZEUS and the supplement 23 to the P series (P.Sup23) of ITU-T Recommendations. The NOIZEUS database is used in the assessment of 11 speech enhancement algorithms whereas the P.Sup23 database is used in the ITU-T 8 kbit/s codec (Recommendation G.729) characterization test. The performance of the proposed speech quality measure is also compared to the results from some traditional objective quality measures. In general, the proposed neural-response-based metric yielded better results than most of the traditional acoustic-property-based quality measures. The proposed metric can be applied to evaluate the performance of various speech-enhancement algorithms and compression systems. [ABSTRACT FROM AUTHOR]

Low bit-rate speech codec offers so many advantages over other codecs that it has become increasingly popular in audio communications such as mobile and Voice over Internet Protocol (VoIP) communications, and thus, researching steganography in low bit-rate speech codec is of important significance. In this study, we proposed a universal VoIP steganography model for low bit-rate speech codec that uses the PESQ deterioration rate and the decoding error to automatically choose a data-embedding algorithm for each VoIP bitstream, which enables ones to achieve covert communications using a low bitrate speech codec efficiently and securely. Because no or little attention has been paid to steganography in Internet Speech Audio Codec, it was chosen as the test codec to verify the effectiveness, security, and practicability of the proposed steganography model. The experimental results show that, with the proposed steganography model, it achieved the average PESQ deterioration rate of 4.04% (less than 5%, indicating strong imperceptibility) and a high data-hiding capacity up to 12 bits/frame (400 bits/s, three times larger than other methods), and the proposed steganography model could effectively resist the latest steganalysis. [ABSTRACT FROM AUTHOR]

This paper presents a method for detection of adaptive multirate (AMR) audio steganography. AMR audio codec is an audio data compression scheme optimized for speech coding, and widely used in some mobile telecommunications system. The AMR audio steganography schemes are emerging recently and they embed secret messages by modifying the nonzero pulse positions which are determined by fixed codebook search in AMR compression procedure. Those methods have high embedding capacity and good imperceptivity. We have observed that those steganography schemes will cause the probability of same pulse positions in the same track increasing. Based on this phenomenon, this paper presents a set of steganalysis features of the probability of same pulse position. The support vector machine is applied to the proposed features and used as the steganalyzer. The performance of the scheme is tested on a database containing $\sim 140$ 714 audios. Experimental results show that the correct detection rate of our proposed method is >90% when the embedding bit rate is 30% or above, and can reach above 85% for cover audios. [ABSTRACT FROM PUBLISHER]

Speech coding has been major issue in the area of digital speech processing. Speech coding is the process of transforming the speech signal in a more compressed form, which can then be transmitted with few numbers of binary digits. It is not possible to access unlimited bandwidth of a channel each time we send a signal across it which leads to code and compress speech signals. Speech compression is applied in long distance communication, high-class speech storage, and message encryption. Speech coding is a lossy type of coding and hence the output signal does not exactly sound like the input. Speech coding techniques discussed here are Linear predictive coding, waveform coding, Code excited linear predictive coding, etc. Linear Predictive Coding and Code Excited Linear Predictive Coding techniques are studied with the help of MATLAB to check their performance measures like compression ratio and speech audible quality. [ABSTRACT FROM AUTHOR]

The typical approach to steganography is to compress the covert data in order to limit its size, which is reasonable in the context of a limited steganographic bandwidth. Trancoding steganography (TranSteg) is a new IP telephony steganographic method that was recently proposed that offers high steganographic bandwidth while retaining good voice quality. In TranSteg, compression of the overt data is used to make space for the steganogram. In this paper we focus on analyzing the influence of the selection of speech codecs on hidden transmission performance, that is, which codecs would be the most advantageous ones for TranSteg. Therefore, by considering the codecs which are currently most popular for IP telephony we aim to find out which codecs should be chosen for transcoding to minimize the negative influence on voice quality while maximizing the obtained steganographic bandwidth. [ABSTRACT FROM AUTHOR]

This work aims to present a combined version of reduced candidate mechanism (RCM) and iteration‐free pulse replacement (IFPR) as a novel and efficient way to enhance the performance of algebraic codebook search in an algebraic code‐excited linear‐prediction speech coder. As the first step, individual pulse contribution in each track is given by RCM, and the value of N is then specified. Subsequently, the replacement of a pulse is performed through the search over the sorted top N pulses by IFPR, and those of 2–4 pulses are carried out by a standard IFPR. Implemented on a G.729A speech codec, this proposal requires as few as 20 searches, a search load tantamount to 6.25% of G.729A, 31.25% of the global pulse replacement method (iteration = 2), 41.67% of IFPR, but still provides a comparable speech quality in any case. The aim of significant search performance improvement is hence achieved in this work. [ABSTRACT FROM AUTHOR]

A method was proposed to skip the look-ahead samples in this paper. Firstly, the method used the covariance algorithm to calculate the linear prediction (LP) coefficients. Secondly, LP coefficients were utilized to produce the new samples. Finally, the new samples were used to replace the look-ahead samples. To evaluate the voice quality, the objective evaluation and subjective evaluation were designed respectively and the complexity test was calculated. Experiments show that the reconstructed quality of the proposed method is near to the original AMR codec, and the delay of the proposed method is lower 5 ms than the original method. [ABSTRACT FROM AUTHOR]

This letter proposes a bandwidth extension (BWE) method using the cepstral envelope coding and duplication of quantized wideband (WB) signals by means of analysis -by-synthesis (AbS) for super-wideband (SWB) coders. In the proposed method, a high frequency band is generated by utilizing the quantized cepstral coefficients extracted from the envelope and the quantized modified discrete cosine transform (MDCT) shape of the wideband signal. The proposed method is compared with the latest G.718 SWB codec and the experimental results show that the proposed method outperforms the baseline codec both in subjective listening tests and objective performance measures. [ABSTRACT FROM AUTHOR]

In recognition of high-quality wideband speech codecs, several standardization activities have been conducted, resulting in the selection of a wideband speech codec called adaptive multi-rate wideband (AMR-WB). The algebraic code-excited linear prediction (ACELP) technique is recommended in AMR-WB, and it is noted that most of the complexity in the ACELP structure comes from the codebook search. In this paper, a new method is proposed for codebook search based on the behavior of backward filtered target signal, d( n), introduced in ITU-T G.722.2 recommendation. To optimize the proposed scheme, five optimization algorithms (i.e., modified genetic algorithm, particle swarm optimization with dynamic inertia weight, bee colony optimization, modified differential evolution, and imperialist competition algorithm) are investigated. Experimental results show that the reduction in codebook search operations of the proposed method is able to reach up to 59 percent as compared with ITU-T G.722.2 recommendation. Meanwhile, BCO-based codebook search scheme has better convergence speed without significant degradation in quality metrics, such as segmental signal-to-noise ratio, mean opinion score, and perceptual evaluation of speech quality, when used in an AMR-WB speech codec. [ABSTRACT FROM AUTHOR]

The memory space and bandwidth may be limited in speech signal communication. To solve the problem, a low rate speech coding and decoding system has been developed to realize speech communication with a high condensation. This system is composed of micro-controller STM32, special speech processor TR600, speech codec chip TLV320AIC10, speech acquisition and playback circuit. This design can code and decode speech in 0. 6 Kbps, 0. 8 Kbps, 1. 2 Kbps and 2. 4 Kbps. The tests show that the system' s reconstructive speech could maintain high naturalness, articulation and intelligibility, hence it reflects the design is feasible, stable and valuable. [ABSTRACT FROM AUTHOR]

Voice over Internet Protocol (VoIP) applications based on peer-to-peer (P2P) communications have been experiencing considerable growth in terms of number of users. To overcome filtering policies or protect the privacy of their users, most of these applications implement mechanisms such as protocol obfuscation or payload encryption that avoid the inspection of their traffic, making it difficult to identify its nature. The incapacity to determine the application that is responsible for a certain flow raises challenges for the effective management of the network. In this paper, a new method for the identification of VoIP sessions is presented. The proposed mechanism classifies the flows, in real-time, based on the speech codec used in the session. To make the classification lightweight, the behavioral signatures for each analyzed codec were created using only the lengths of the packets. Unlike most previous approaches, the classifier does not use the lengths of the packets individually. Instead, it explores their level of heterogeneity in real time, using entropy to emphasize such feature. The results of the performance evaluation show that the proposed method is able to identify VoIP sessions accurately and simultaneously recognize the used speech codec. [ABSTRACT FROM AUTHOR]

In this letter, we propose a simple and efficient method, called `Dual-CRC', to improve speech quality based on the list Viterbi algorithm (LVA) in the Universal Mobile Telecommunications System (UMTS). We also employ the well-known joint source-channel decoding (JSCD) scheme together with LVA decoding to further improve the speech quality. Simulation results show that the two proposed decoders, the Dual-CRC LVA and the combination of the LVA and the JSCD, can significantly improve the speech quality for UMTS, especially in low signal-to-noise ratio and low mean opinion score regions. [ABSTRACT FROM PUBLISHER]

This paper presents, optimized wavelet filters using windowing techniques and linear optimization, which are used as mother wavelets for speech compression based on wavelets. Proposed design and real-time implementation of the optimized speech codec based on discret wavelet transform (DWT), was performed. The comparison tests between the proposed optimized wavelet filters and others wavelets based on CR, SNR, PSNR and NRMSE computation, confirm the efficiency of the developed wavelet filters. The real-time implementation using DSP/BIOS tool, on the TMS320C6416 fixed-processor shows that the encoding and decoding algorithms satisfy the real-time processing. [ABSTRACT FROM AUTHOR]

Low bit-rate speech codecs have been widely used in audio communications like VoIP and mobile communications, so that steganography in low bit-rate audio streams would have broad applications in practice. In this paper, the authors propose a new algorithm for steganography in low bit-rate VoIP audio streams by integrating information hiding into the process of speech encoding. The proposed algorithm performs data embedding while pitch period prediction is conducted during low bit-rate speech encoding, thus maintaining synchronization between information hiding and speech encoding. The steganography algorithm can achieve high quality of speech and prevent detection of steganalysis, but also has great compatibility with a standard low bit-rate speech codec without causing further delay by data embedding and extraction. Testing shows, with the proposed algorithm, the data embedding rate of the secret message can attain 4 bits/frame (133.3 bits/second). [ABSTRACT FROM PUBLISHER]

This paper has designed a cross-layer framework for voice over Internet protocol (VoIP) services in IEEE 802.16 systems. It uses the application session information of the session description protocol to generate the quality of service parameters in IEEE 802.16 systems. This feature allows the system to efficiently allocate the radio resource because it can exactly estimate the properties of VoIP services such as packet-size and packet-generation-interval. In other words, the cross-layer framework is expected to achieve a novel resource request scheme for a VoIP service that dynamically assigns the radio resource. This paper has analyzed the maximum number of supportable VoIP users for the resource request schemes in terms of the packet-generation-interval in the silent-period, the duration of the silent-period, and the major VoIP speech codec. The numerical results show that the proposed scheme can efficiently support the VoIP services for the various communication environments. Particularly, it can improve the maximum number of supportable VoIP users by 14\sim 93\% compared to an extended real-time polling service. [ABSTRACT FROM PUBLISHER]

The adaptive multi-rate wideband (AMR-WB) speech codec with a sampling rate of 16 kHz is one of the speech codecs applied to 4G mobile communication systems as a way to remarkably improve the speech quality of a smartphone. However, a major drawback is that the vector quantization of the immittance spectral frequency (ISF) coefficients takes up the second largest of the total computational load of the AMR-WB encoder. In other words, the speech quality is improved at the cost of high battery power consumption. Accordingly, this letter presents a triangular inequality elimination algorithm equipped with a dynamic and an intersection mechanism, named DI-TIE, as a means to considerably improve the performance of ISF coefficient quantization in AMR-WB speech codec. Both mechanisms are designed in a way that recursively enhances the performance of TIE algorithm. This proposal is demonstrated by experimental results as a superior search algorithm relative to a conventional TIE and a multiple TIE (MTIE) approaches. With a full search algorithm as a benchmark for search load comparison, this work provides a search load reduction of more than 77%, a figure far beyond the 36% in the TIE and 49% in the MTIE approach. © 2016 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc. [ABSTRACT FROM AUTHOR]

The quality of time sensitive interactive voice traffic can seriously degrade if moved over lossy wired or wireless IP networks, particularly with low rate speech. One technique that can be used to improve the quality is to transmit interleaved, and possibly duplicated, voice frames. This paper compares the quality of non-interleaved clear MELP voice traffic with the quality of several interleaving schemes over a high loss packet network. Multiple NATO languages were tested. The results show that interleaving can significantly improve the perceived voice quality, but at the expense of an increase in mouth-to-ear delivery delay and, if redundancy is enabled, an increase in required bandwidth. The improved voice quality can be quite significant at high packet loss rates. [ABSTRACT FROM AUTHOR]