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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]

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]

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]

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]

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]

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]

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]

The authors present and evaluate a technique for synchronizing the internal states of a code-excited-linear-prediction (CELP) encoder and decoder after the occurrence of frame erasure. The designed technique, called ldquoduplicated transmission (DT),rdquo uses some redundant information for realizing synchronization. The encoder performs encoding processes twice and sends two codes for each frame. One code is encoded by an encoder that is initialized. The code is used in cases where the previous frame is erased. An onset detector is combined with the DT technique to select the frames to which the DT should be applied. Subjective test results suggest that, by introducing DT selectively, the number of DT frames is reducible by about 80% without degrading the subjective quality. Results demonstrate that synchronization of the internal states is effective in cases of erasure of onset. The DT technique requires no additional algorithmic delay. For that reason, it would a better choice for particular applications for which the delay has a significant impact. [ABSTRACT FROM PUBLISHER]

Network-based speech recognition (NSR) and distributed speech recognition (DSR) have been proposed as solutions to translate speech recognition technologies to mobile environments. NSR is the most straightforward solution since it does not require any modification in the mobile phone, however DSR offers higher robustness against codec compression and transmission channel degradation. This paper explores an alternative approach for remote speech recognition which combines the advantages of NSR and DSR. In this scheme, a standard speech codec is used for speech transmission but the recognition is performed from the received codec parameters. In particular, we focus on the effect of transmission channel errors, which can cause a more severe performance reduction on speech recognition than codec distortion. First, we show that an NSR solution can approach DSR through a reconstruction technique along with an adapted noise reduction technique originally proposed for acoustic noise. Then, these results are improved by working with recognition features directly extracted from the codec bitstream by means of parameter transcoding. Required modifications on current networks in order to access the bitstream are described. The network upgrading with the tandem free operation (TFO) protocol is an attractive solution. This upgrade not only offers an overall improvement on the end-to-end speech quality, but would also allow a recognition performance similar, and even higher in poor channel conditions, to that obtained by DSR when parameter transcoding along with the proposed mitigation techniques are applied [ABSTRACT FROM PUBLISHER]

Voice is the preferred method of human communication. Although there have been times when it seemed that the voice communications problem was solved, such as when the PSTN was our primary network or later when digital cellular networks reached maturity, such is not the case today. This paper addresses the challenges and opportunities starting from the basic issues in speech coder design, developing the important speech coding techniques and standards, discussing current and future applications, outlining techniques for evaluating speech coder performance, and identifying research directions. The most prominent speech coding standards are presented and their properties, such as performance, complexity, and coding delay, analyzed. Particular networks and applications for each standard are included. Further, reflecting upon the issues and developments highlighted in this paper, it becomes evident that there is a diverse set of challenges and opportunities for research and innovation in speech coding and voice communications. [ABSTRACT FROM PUBLISHER]

A new system enhancement method is proposed for the EIA/TIA-136 system offering both channel operational range extension and improved performance within the current operational range. The existing time-division multiple-access (TDMA) (136) speech codec, the IS-641 enhanced full rate vocoder, operates at a fixed bit rate and does not allow the reallocation of bits to channel error protection as channel conditions degrade. The research presented here investigates the application of the narrow-band adaptive multirate (NB-AMR) speech codec and the wide-band AMR (WB-AMR) codec, both originally designed for the 200 kHz GSM channel, in the TDMA (TIA/EIA-136) 30-kHz system. In particular, we investigate adaptively allocating bits between NB/WB speech coding and error control coding within the limited channel bandwidth. Four modes out of 17 have been carefully chosen for the new TDMA/AMR system. Switching between codec rates as channel conditions change produces range extension below a C/I of 15 dB while also improving performance in the existing operational range above 15 dB. We keep the time slot formats unchanged so that our method is completely compatible with existing 136 systems. [ABSTRACT FROM PUBLISHER]

This article reviews state-of-the-art in transport adaptation techniques for mobile networks. It discusses the mechanisms for rate adaptation to combat quality degradations of speech caused by the radio links. It begins with a review of dynamic schemes for adaptation of speech encoders in cellular networks where we observe two distinct approaches to rate adaptation: network controlled and source controlled. The issues associated with adaptive voice over IP (VoIP) mechanisms are considered next. Here, the encoder detects some form of network congestion to judge how to behave itself for the good of the network. It is noted that this altruistic behavior will only benefit coordinated IP networks such as private intranets and its application to the public Internet is improbable. [ABSTRACT FROM PUBLISHER]

Future wireless multimedia terminals will have a variety of applications that require speech recognition capabilities. We consider a robust distributed speech recognition system where representative parameters of the speech signal are extracted at the wireless terminal and transmitted to a centralized automatic speech recognition (ASR) server. We propose two unequal error protection schemes for the ASR bit stream and demonstrate the satisfactory performance of these schemes for typical wireless cellular channels. In addition, a "soft-feature" error concealment strategy is introduced at the ASR server that uses "soft-outputs" from the channel decoder to compute the marginal distribution of only the reliable features during likelihood computation at the speech recognizer. This soft-feature error concealment technique reduces the ASR error rate by more than a factor of 2.5 for certain channels. Also considered is a channel decoding technique with source information that improves ASR performance [ABSTRACT FROM PUBLISHER]

The processing delay is a serious constraint for speech communication. A one-way end-to-end delay of more than 150 ms can severely degrade the quality of real-time conversations. The components of the total system delay includes the speech frame size, the look ahead, other algorithmic delays, multiplexing delay, processing delay for computation, and transmission delay. At the transcoder rate adaptor unit (TRAU), the only delay that can be manipulated is the processing delay. The TRAUs are generally positioned remote to the base transceiver station (BTS). The channel codec units (CCUs) are located in the BTS. In general, 16 kbit/s traffic channels can be used for full rate speech between the TRAU and BTS. By putting the TRAU remote to the BTS, DSPs for speech coding can be utilized more efficiently to cut system cost. High performance DSPs, such as the Lucent DSP16000, can be used to further cut the cost per speech channel. This article presents an implementation of GSM enhanced full rate (EFR) codec on the Lucent Technologies' DSP16000. The original European Telecommunications Standards Institute (ETSI) C code has been restructured to address the issues of MIPS (million instructions per second), RAM usage, and processing delay. We give a performance overview of vocoder implementations on some existing fixed-point DSPs and discuss the architecture of the DSP16000. Details on how the ETSI C code is restructured are presented. The DSP16210 implementation results are then discussed [ABSTRACT FROM PUBLISHER]

Speech coding is a well researched area, and researchers are making new proposals for im provements to current algorithms. The family of code-excitated linear prediction coders rep resents a recent breakthrough, which almost caters to the present need for low bit rates. In mobile communications, low bit-rate speech coders play a crucial role in spectrally efficient transmission. Theoretically, a rate of a few hundred bits per second is enough to code speech efficiently. Many coders have been de veloped using computationally intensive tech niques that can achieve rates of 4 to 6 kilobits per second. Theoretically, there is room for improvement. The trend is to use analytical models of speech production and perception. This paper briefly reviews existing speech cod ers, identifies future trends in this research area and describes results obtained from simu lating vowels and consonant waveforms and from a multi-pulse excitation/linear predictive speech coders algorithm. [ABSTRACT FROM PUBLISHER]