Wideband Speech Codec Research Articles

Convolutional Neural Networks to Enhance Coded Speech

Enhancing coded speech suffering from far-end acoustic background noise, quantization noise, and potentially transmission errors, is a challenging task. In this work we propose two postprocessing approaches applying convolutional neural networks (CNNs) either in the time domain or the cepstral domain to enhance the coded speech without any modification of the codecs. The time domain approach follows an end-to-end fashion, while the cepstral domain approach uses analysis-synthesis with cepstral domain features. The proposed postprocessors in both domains are evaluated for various narrowband and wideband speech codecs in a wide range of conditions. The proposed postprocessor improves speech quality (PESQ) by up to 0.25 MOS-LQO points for G.711, 0.30 points for G.726, 0.82 points for G.722, and 0.26 points for adaptive multirate wideband codec (AMR-WB). In a subjective CCR listening test, the proposed postprocessor on G.711-coded speech exceeds the speech quality of an ITU-T-standardized postfilter by 0.36 CMOS points, and obtains a clear preference of 1.77 CMOS points compared to legacy G.711, even better than uncoded speech with statistical significance. The source code for the cepstral domain approach to enhance G.711-coded speech is made available.

An Upgraded Version of the Binary Search Space-Structured VQ Search Algorithm for AMR-WB Codec

Adaptive multi-rate wideband (AMR-WB) speech codecs have been widely used for high speech quality in modern mobile communication systems, e.g., handheld mobile devices. Nevertheless, a major handicap is that a remarkable computational load is required in the vector quantization (VQ) of immittance spectral frequency (ISF) coefficients of an AMR-WB coding. In view of this, a two-stage search algorithm is presented in this paper as an efficient way to reduce the computational complexity of ISF quantization in AMR-WB coding. At stage 1, an input vector is assigned to a search subspace in an efficient manner using the binary search space-structured VQ (BSS-VQ) algorithm, and a codebook search is performed over the subspace at stage 2 using the iterative triangular inequality elimination (ITIE) approach. Through the use of the codeword rejection mechanisms equipped in both stages, the computational load can be remarkably reduced. As compared with the original version of the BSS-VQ algorithm, the upgraded version provides a computational load reduction of up to 51%. Furthermore, this work is expected to satisfy the energy saving requirement when implemented on an AMR-WB codec of mobile devices.

A scalable wideband speech codec using the wavelet packet transform based on the internet low bitrate codec

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.

Automatic speaker verification on narrowband and wideband lossy coded clean speech

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.

An Efficient VQ Codebook Search Algorithm Applied to AMR-WB Speech Coding

The adaptive multi-rate wideband (AMR-WB) speech codec is widely used in modern mobile communication systems for high speech quality in handheld devices. Nonetheless, a major disadvantage is that vector quantization (VQ) of immittance spectral frequency (ISF) coefficients takes a considerable computational load in the AMR-WB coding. Accordingly, a binary search space-structured VQ (BSS-VQ) algorithm is adopted to efficiently reduce the complexity of ISF quantization in AMR-WB. This search algorithm is done through a fast locating technique combined with lookup tables, such that an input vector is efficiently assigned to a subspace where relatively few codeword searches are required to be executed. In terms of overall search performance, this work is experimentally validated as a superior search algorithm relative to a multiple triangular inequality elimination (MTIE), a TIE with dynamic and intersection mechanisms (DI-TIE), and an equal-average equal-variance equal-norm nearest neighbor search (EEENNS) approach. With a full search algorithm as a benchmark for overall search load comparison, this work provides an 87% search load reduction at a threshold of quantization accuracy of 0.96, a figure far beyond 55% in the MTIE, 76% in the EEENNS approach, and 83% in the DI-TIE approach.

An efficient search algorithm for ISF vector quantization in AMR‐WB speech codec

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.

Bandwidth Extension of Speech Signals: A Comprehensive Review

Telephone systems commonly transmit narrowband (NB) speech with an audio bandwidth limited to the traditional telephone band of 300-3400 Hz. To improve the quality and intelligibility of speech degraded by narrow bandwidth, researchers have tried to standardize the telephonic networks by introducing wideband (50-7000 Hz) speech codecs. Wideband (WB) speech transmission requires the transmission network and terminal devices at both ends to be upgraded to the wideband that turns out to be time-consuming. In this situation, novel Bandwidth extension (BWE) techniques have been developed to overcome the limitations of NB speech. This paper discusses the basic principles, realization, and applications of BWE. Challenges and limitations of BWE are also addressed.

Effect of Speech Compression on the Automatic Recognition of Emotions

This paper investigates the effects of standard speech compression techniques on the accuracy of automatic emotion recognition. Effects of Adaptive Multi-Rates (AMR), Adaptive Multi-Rate Wideband (AMR-WB) and Extended Adaptive Multi-Rate Wideband (AMR-WB+) speech codecs were compared against emotion recognition from uncompressed speech. The recognition methods included techniques based on three different types of acoustic speech parameters: Teage Energy Operator features (TEO), Mel Frequency Cepstral Coefficients (MFCCs), and Glottal Time and Frequency domain features (GP-T and GP-F). The results showed that in general, all three speech compression techniques resulted in the reduction of emotion recognition accuracy. However, the amount of degradation varied across compression methods and types of acoustic features. It was observed that the accuracy of emotion recognition using the AMR-WB technique was higher than the accuracy of the AMR-WB+ and the AMR codecs. Further, the TEO-PWP features showed much more robust performance under different compression rates than the MFCC, GP-T and GP-F features.

Modification of codebook search in adaptive multi-rate wideband speech codecs using intelligent optimization algorithms

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.

Investigation of Automatic Speech Recognition Performance and Mean Opinion Scores for Different Standard Speech and Audio Codecs

Usage of Automatic Speech Recognition (ASR) systems is increasing day-by-day for voice centric applications in mobile handheld and Voice over Internet Protocol (VoIP) devices. The necessity is also increasing to find out the ASR performance under different network impediments. Among them, speech and audio coding standards is the one, which affects the ASR performance greatly, when, using them with different sampling and bit rates in the practical systems. Another common impediment which influences the ASR accuracy is the bit errors in the wireless networks and packet drop conditions in the VoIP networks. ASR performance with some of the speech coding standards under noise conditions for the wireless networks is reported in the literature. However, each study is reporting the ASR performance for few narrowband codecs with different speech databases and different ASR toolkits like RAPHEL, HTK, SPHINX, etc. In this paper, the analysis on ASR performance while using both narrowband and wideband speech and audio coding standards, which are currently accepted for GSM mobile and VoIP networks, using the common speech database-TIMIT, and using ASR toolkit-SPHINX, is presented. The Mean Opinion Score (MOS), which is the generally accepted speech quality measurement technique, is also analyzed for all the speech and audio coding standards, using the same speech database. The results of the studies carried out for the ASR word accuracies and MOS values for different narrowband and wideband speech and audio codecs under no-loss conditions are presented. Results for different rates of packet drop condition which is the common noise scenario in wired networks such as VoIP (which is also merging with wireless networks) are also presented. The observation is that though some of the codecs are showing poor MOS performance at lower bit rates, the corresponding ASR performance is comparable with other codecs at higher bit rates.

Real-Time Implementation AMR-WB Algorithm in TMS320VC5509A DSP

Adaptive multi-rate wideband (AMR-WB) speech codec is the first standard for both wired and wireless universal broadband speech codec. Real-time implementation of AMR-WB algorithm in DSP has important practical significance and will accelerate its course of practice. The paper first designed the hardware platform and system software structure for real-time implementation of the AMR-WB algorithm, then introduced the transplantation of AMR-WB algorithm to the designed hardware system. The paper next discussed the real-time implementation of optimized algorithm on the hardware platform. Objective test results indicated the decoded speeches of real-time platform were bit exactly the same as test sequences. Synthetic speeches of real-time platform were very good in restoring the speaker’s voice characteristics by good intelligibility and naturalness.

정규화 코드북을 이용한 분할 벡터 구조의 ISF 적응적 양자화 기법

In most of the ISF (or LSF) based real time speech codec, SVQ (split vector quantization) method is used to decrease the quantizer complexity and memory size of codebook. However, it produces drawback that the level of correlation between code vectors can not be used during vector splits. This paper presents a new method of adaptive ISF vector quantization, which compensates the drawbacks of SVQ structured quantizer for wideband speech codec. In each different frame, the proposed method makes use of the correlation between splitted vectors by adaptively changing codebook distribution according to ordering property of ISF. The algorithm is evaluated in AMR-WB, and shows about 1.5 bit per frame improvement.

적응적인 확장된 코드북을 이용한 분할 벡터 양자화기 구조의 ISF 양자화기 개선

본 논문에서는 ISF 계수의 순서화 성질을 이용하여 분할구조 벡터양자화기의 단점을 보완하여 ISF 계수 양자화의 성능을 높이는 알고리듬을 제안하고, 이를 이용한 광대역 음성 부호화기용 ISF 계수 양자화기를 설계한다. 16차 이상의 광대역 코덱의 ISF 계수는 계산량과 메모리 사용을 줄이기 위해서 분할구조의 벡터 양자화기를 사용한다. 분할구조 양자화기는 ISF 계수간의 상관도를 충분히 활용하지 못하는 단점이 발생한다. 제안하는 알고리듬은 이러한 단점을 극복하기 위하여 ISF 계수의 순서화 성질을 이용한다. ISF 계수의 순서화 성질을 이용하여 각 서브벡터의 불필요한 코드북 (Codebook Redundancy)을 검색할 수 있다. 이러한 불필요한 코드북은 ISF 계수의 순서화 성질, ISF 계수 예측과정과 기존 코드북의 보간법 (Interpolation)을 통해 적응적인 확장된 코드북으로 교체되어 양자화기의 성능을 향상시킨다. 제안된 알고리듬은 기존의 분할구조 양자화기에서 사용되지 못했던 17 %가량의 불필요한 코드북 인덱스를 적응적인 확장된 코드북에 할당하여, 표준화된 코덱인 AMR-WB의 ISF 계수 양자화기에 비해서 주파수 왜곡 관점에서 약 2 bit 가량의 이득을 보는 결과를 얻었다. This paper presents a method for improving the performance of ISF coefficients quantizer through compensating the defect of the split structure vector quantization using the ordering property of ISF coefficients. And design the ISF coefficients quantizer for wideband speech codec using proposed method. The wideband speech codec uses split structure vector quantizer which could not use the correlation between ISF coefficients fully to reduce complexity and the size of codebook. The proposed algorithm uses the ordering property of ISF coefficients to overcome the defect. Using the ordering property, the codebook redundancy could be figured out. The codebook redundancy is replaced by the adaptive-extended codebook to improve the performance of the quantizer through using the ordering property, ISF coefficient prediction and interpolation of existing codebook. As a result, the proposed algorithm shows that the adaptive-extended codebook algorithm could get about 2 bit gains in comparison with the existing split structure ISF quantizer of AMR-WB (G.722.2) in the points of spectral distortion.

Instrumental Estimation of E-Model Parameters for Wideband Speech Codecs

A method is described for quantifying the quality of wideband speech codecs. Two parameters are derived from signal-based speech quality model estimations: (i) a wideband equipment impairment factor Ie,WB and (ii) a wideband packet-loss robustness factor Bpl, WB. The equipment impairment factor can be combined with impairment factors for other quality degradations to form an estimate of the overall conversational quality R of a wideband communication scenario, using a wideband extension of the E-model. The packet-loss robustness factor captures the robustness of the codec against packet-loss degradations. In contrast to past work, these parameters are no longer determined on the basis of auditory test results, but from signal-based speech quality models. We applied three intrusive models to several databases and compared the derived quality estimates and impairment factors to those obtained from auditory tests. The results show that when migrating from narrowband to wideband transmission--a quality improvement of roughly 30% can be obtained, which is very similar to the one observed in auditory tests. The estimated impairment factors show a high correlation to those derived from auditory scores. Congruences and discrepancies to auditory test results are discussed, and an outline of work necessary to set up a wideband or even superwideband E-model is given.

ITU-T G.711.1: extending G.711 to higher-quality wideband speech

In March 2008 the ITU-T approved a new wideband speech codec called ITU-T G.711.1. This Recommendation extends G.711, the most widely deployed speech codec, to 7 kHz audio bandwidth and is optimized for voice over IP applications. The most important feature of this codec is that the G.711.1 bitstream can be transcoded into a G.711 bitstream by simple truncation. G.711.1 operates at 64, 80, and 96 kb/s, and is designed to achieve very short delay and low complexity. ITU-T evaluation results show that the codec fulfils all the requirements defined in the terms of reference. This article presents the codec requirements and design constraints, describes how standardization was conducted, and reports on the codec performance and its initial deployment.

Improved voice activity detection algorithm using wavelet and support vector machine

This paper proposes an improved voice activity detection (VAD) algorithm using wavelet and support vector machine (SVM) for European Telecommunication Standards Institution (ETSI) adaptive multi-rate (AMR) narrow-band (NB) and wide-band (WB) speech codecs. First, based on the wavelet transform, the original IIR filter bank and pitch/tone detector are implemented, respectively, via the wavelet filter bank and the wavelet-based pitch/tone detection algorithm. The wavelet filter bank can divide input speech signal into several frequency bands so that the signal power level at each sub-band can be calculated. In addition, the background noise level can be estimated in each sub-band by using the wavelet de-noising method. The wavelet filter bank is also derived to detect correlated complex signals like music. Then the proposed algorithm can apply SVM to train an optimized non-linear VAD decision rule involving the sub-band power, noise level, pitch period, tone flag, and complex signals warning flag of input speech signals. By the use of the trained SVM, the proposed VAD algorithm can produce more accurate detection results. Various experimental results carried out from the Aurora speech database with different noise conditions show that the proposed algorithm gives considerable VAD performances superior to the AMR-NB VAD Options 1 and 2, and AMR-WB VAD.

Iterative AMR-WB Source and Channel Decoding Using Differential Space–Time Spreading-Assisted Sphere-Packing Modulation

In this paper, we present a novel system that invokes jointly optimized iterative source and channel decoding for enhancing the error resilience of the adaptive multirate wideband (AMR-WB) speech codec. The resultant AMR-WB-coded speech signal is protected by a recursive systematic convolutional (RSC) code and transmitted using a noncoherently detected multiple-input-multiple-output (MIMO) differential space-time spreading (DSTS) scheme. To further enhance the attainable system performance and to maximize the coding advantage of the proposed transmission scheme, the system is also combined with multidimensional sphere-packing (SP) modulation. Furthermore, the convergence behavior of the proposed scheme is evaluated with the aid of extrinsic information transfer (EXIT) charts. The proposed system exhibits an <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">Eb</i> / <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</i> <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">0</sub> gain of about 1 dB, as compared with the benchmark scheme carrying out joint channel decoding and DSTS-aided SP demodulation in conjunction with separate AMR-WB decoding, when using only <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">I</i> <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">system</sub> = 2 system iterations and when communicating over narrow-band correlated Rayleigh fading channels.

Block Constrained Trellis Coded Vector Quantization of LSF Parameters for Wideband Speech Codecs

In this paper, block constrained trellis coded vector quantization (BC-TCVQ) is presented for quantizing the line spectrum frequency parameters of the wideband speech codec. Both a predictive structure and a safety-net concept are combined into BC-TCVQ to develop the predictive BC-TCVQ. The performance of this quantization is compared with that of the linear predictive coding vector quantizer used in the AMR-WB codec, demonstrating reductions in spectral distortion.

Packet Loss Concealment for G.722 using Side Information with Application to Voice over Wireless LANs

The G.722 wideband speech codec offers higher quality and better naturalness than G.711, is low in com- plexity, has low delay, and tandems well with other codecs. This makes it an attractive codec for Voice over IP and Voice over Wireless LANs. However, loss of a G.722 coded speech frame results in a mismatch of the encoder/decoder states that affects the decoding of subsequent correctly received frames. We present a new packet loss concealment (PLC) method for G.722 and propose an efficient approach to sending the side information to improve concealment of the lost frame and to resynchronize the encoder and decoder. The proposed method greatly improves the robustness of the G.722 codec to packet losses for Voice over Wireless LANs. Wideband Perceptual Evaluation of Speech Quality Mean Opinion Score (WB-PESQ MOS) is used as the quality metric for evaluating the performance of the PLC schemes.

Wideband Speech Coding Advances in VMR-WB Standard

This paper presents novel techniques for source-controlled variable-rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by the Third-Generation Partnership Project 2 (3GPP2) for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third-generation wireless system. The codec utilizes efficient coding modes optimized for different classes of speech signal including generic coding based on AMR-WB for transients and onsets, voiced coding optimized for stable voiced signals, unvoiced coding optimized for unvoiced segments, and comfort noise generation for inactive segments. Several innovations enable very good performance at average bit rates below 8 kb/s for active speech coding. The article presents an overview of the codec and describes in detail some of the codec novel features: Robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Selected results from the Selection and Characterization tests of the codec illustrate its performance