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wideband speech coding advances in VMR-WB standard

IEEE TRANSACTIONS ON AUDIO speech AND LANGUAGE PROCESSING 2007年第4期15卷 1167-1179页

作者： Jelinek, Milan Salami, Redwan Univ Sherbrooke Dept Elect & Comp Engn Sherbrooke PQ J1K 2R1 Canada VoiceAge Corp Montreal PQ H3R 2H6 Canada

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.

关键词： linear predictive coding standardization variable-rate speech coding wideband speech coding

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Waveform interpolation for 4 kbit/s wideband speech coding

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ELECTRONICS LETTERS 2002年第10期38卷 486-487页

作者： Ritz, CH Burnett, IS Univ Wollongong Whisper Labs Wollongong NSW 2522 Australia

An implementation of waveform interpolation (WI) applied to wideband speech coding at 4 kbit/s is presented. Listening tests comparing narrowband speech coded using standard coders are presented, with wideband speech coded using WI. Results show a clear preference for wideband speech over narrowband speech coded at an equivalent bit rate.

关键词： subjective listening tests interpolation speech coding waveform interpolation waveform analysis Interpolation and function approximation (numerical analysis) speech and audio coding wideband speech coding 4 kbit/s narrowband speech

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MDCT-Domain Packet Loss Concealment for Scalable wideband speech coding

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2nd International Conference on Ubiquitous Computing and Multimedia Applications

作者： Park, Nam In Kim, Hong Kook Gwangsju Inst Sci & Technol GIST Sch Informat & Commun Kwangju 500712 South Korea

ISBN: (纸本)9783642209970

In this paper, we propose a modified discrete cosine transform (MDCT) based packet loss concealment (PLC) algorithm in order to improve the quality of decoded speech when a packet loss occurs in scalable wideband speech coders using MDCT as spectral parameters. The proposed PLC algorithm is realized by smoothing MDCT coefficients between the low and high bands for scalable wideband speech coders. In G.729.1, a typical scalable wideband speech coder standardized by ITU-T, two different PLC algorithms are applied to low band and high band in time and frequency domain, respectively. Thus, the MDCT coefficients around the boundary between the low and high band can be mismatched. The proposed PLC algorithm is replaced with the PLC algorithm applied to the high band, and it compensates for the mismatch in the MDCT domain at the boundary. Finally, we compare the performance of the proposed PLC algorithm with that of the PLC algorithm employed in G.729.1 by means of perceptual evaluation of speech quality (PESQ), an A-B preference test, and a waveform comparison under different random and burst packet loss conditions. It is shown from the experiments that the proposed PLC algorithm provides significantly better speech quality than the PLC of G.729.1.

关键词： Packet loss concealment (PLC) wideband speech coding modified discrete cosine transform (MDCT) G.729.1

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Multiband vector quantization based on inner product for wideband speech coding

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IEICE TRANSACTIONS ON INFORMATION AND SYSTEMS 2005年第11期E88D卷 2606-2608页

作者： Chang, JH Mitra, SK Inha Univ Dept Elect Engn Inchon South Korea Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.

关键词： wideband speech coding VQ CELP MLT inner product

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A wideband speech and audio coding candidate for ITU-T G.711WBE standardization

A wideband speech and audio coding candidate for ITU-T G.711...

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33rd IEEE International Conference on Acoustics, speech and Signal Processing

作者： Hiwasaki, Yusuke Mori, Takeshi Sasaki, Shigeaki Ohmuro, Hitoshi Kataoka, Akitoshi NTT Cyber Space Labs Tokyo 1808585 Japan

ISBN: (纸本)9781424414833

This paper describes an ITU-T G.711 embedded wideband speech coder, submitted as a candidate to the ITU-T G.711wideband Extension standardization qualification phase, The codec generates a bitstream comprised of three layers: a G.711 core layer with noise shaping, a time-domain weighted vector quantized narrowband enhancement layer, and an MDCT-based higher band enhancement layer. Through subjective evaluations, the coder was found to meet all tested requirements and objectives set in terms of reference, with a low computational complexity at 10 WMOPS.

关键词： wideband speech coding ITU-TG.711 pulse code modulation

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Intra- and inter-frame prediction in bandwidth scalable coding of wideband speech

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IET SIGNAL PROCESSING 2011年第2期5卷 164-170页

作者： Song, G. -B. Samsung Elect Co Ltd Elect Examinat Div Elect & Elect Examinat Bur Taejon 302701 South Korea

This study discusses a predictive coding scheme that utilises both the intra-frame correlation (here referred to as redundancy between frequency bands) and inter-frame correlation in bandwidth scalable coding of wideband speech. Prediction error filters that combine intra-frame prediction with inter-frame prediction in a cascade connection are examined. Three representative intra-frame functions that have been developed in the area of bandwidth extension (BWE) are considered to see how well they exploit the intra-frame correlation. In addition, some resulting changes are investigated when they are combined with the predictive vector quantiser that exploits the inter-frame correlation. The performances of the Gaussian mixture model-based and hidden Markov model-based BWE functions are similar in the context of intra-frame coding, but both of them significantly outperform the linear BWE function in the same context. However, it is also observed that the linear function is the most promising in the context of the joint predictive coding of intra-frame and inter-frame. These results are supported by some statistical tests.

关键词： hidden Markov model bandwidth scalable coding correlation methods prediction error filters bandwidth extension quantisation (signal) wideband speech coding Filtering methods in signal processing speech coding statistical testing intraframe coding speech and audio coding Gaussian processes linear BWE function hidden Markov models Gaussian mixture model speech processing techniques interframe correlation prediction predictive vector quantiser intraframe correlation prediction filtering theory predictive coding scheme Markov processes statistical tests

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Bandwidth extension for hierarchical speech and audio coding in ITU-T Rec. G.729.1

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IEEE TRANSACTIONS ON AUDIO speech AND LANGUAGE PROCESSING 2007年第8期15卷 2496-2509页

作者： Geiser, Bernd Jax, Peter Vary, Peter Taddei, Herve Schandl, Stefan Gartner, Martin Guillaume, Cyril Ragot, Stephane Univ Aachen Rhein Westfal TH Aachen Inst Commun Syst & Data Proc IND D-52056 Aachen Germany Siemens AG D-80333 Munich Germany Siemens AG A-1210 Vienna Austria France Telecom R&D TECH SSTP F-22307 Lannion France

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated "excitation signal." Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

关键词： bandwidth extension hierarchical bitstream organization wideband speech coding

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Voicing-Based Codebook in Low-Rate wideband CELP coding

Voicing-Based Codebook in Low-Rate Wideband CELP Coding

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Interspeech Conference 2007

作者： Guerchi, Driss Rabie, Tamer Louzi, Abderhani UAE Univ Coll Informat Technol Al Ain U Arab Emirates Ericsson Canada Montreal PQ Canada

ISBN: (纸本)9781605603162

In this paper we propose a new technique to quantize the spectral information in an Algebraic Code-Excited Linear Prediction (ACELP) wideband codec. The Voicing-Based Vector Quantization (V-BVQ) presented in this paper, optimizes the search of the optimum vector while reducing the codebook entries by almost one third. In the VBVQ training phase, three codebooks are individually designed for voiced, unvoiced and transition speech. The proposed technique reduces the processing delay since it restricts the quantization of an input vector to only one of the three codebooks. For each speech frame, one codebook is selected based on the interframe correlation of the spectral information. The VBVQ was successfully implemented in an ACELP wideband coder. ne objective and subjective performance are superior to that of the combination of the split vector quantization and multistage vector quantization after using the same database for training and testing.

关键词： wideband speech coding voicing-based vector quantization low-complexity ISF quantization

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A wideband Scalable Bit Rate Mixed Excitation Linear Prediction-Enhanced speech Coder by Preserving Speaker-Specific Features

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CIRCUITS SYSTEMS AND SIGNAL PROCESSING 2023年第6期42卷 3437-3463页

作者： Sankar, M. S. Arun Sathidevi, P. S. Natl Inst Technol Calicut Dept Elect & Commun Engn Calicut 673601 Kerala India

There has been a significant growth in the mobile devices and services, fuelling an increasing demand for voice-activated applications. In this context, it is important that individual speaker characteristics are captured, in addition to the salient information in the speech signal. Thus, efficient speech coders that can achieve the dual goals of compact speech representation that maintains speech intelligibility and quality, and preservation of speaker-specific characteristics are attractive. A wideband scalable bit rate mixed excitation linear prediction-enhanced speech coder with an efficient representation for excitation using glottal instants and linear predictive coding based on mel scale is proposed in this paper. The instantaneous pitch or epoch is included in the excitation to get an accurate estimation of glottal instants, a vital parameter in speaker recognition. By optimizing the bit requirement using speech category-based coding, the proposed wideband coder can operate at bit rates ranging from 3.3 to 5.1 kbps with an average bit rate of 3.6 kbps. The proposed coder provides, at 3.6 kbps, similar perceptual quality, as measured by mean opinion score and perceptual evaluation of speech quality, as that of code excited linear prediction operating at 6.4 kbps. The performance of the proposed coder in speaker recognition is analysed, and it gives an equal error rate of 12.5%, which is very promising.

关键词： LPC Linear prediction MELP speech coding speech processing wideband speech coding

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Fast recovery for a CELP-Like speech codec after a frame erasure

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IEEE TRANSACTIONS ON AUDIO speech AND LANGUAGE PROCESSING 2007年第8期15卷 2485-2495页

作者： Chibani, Mohamed Lefebvre, Roch Gournay, Philippe Univ Sherbrooke Dept Genie Elect & Genie Informat Sherbrooke PQ J1K 2R1 Canada

The adaptive codebook used in code-excited linear prediction (CELP)-like speech codecs is very effective for modeling the quasi-periodic component of the excitation signal but, unfortunately, introduces a strong interframe dependency that renders the decoder vulnerable to frame erasures. For voiced speech, the error affects not only the erased frame but also all the subsequent frames. In this paper, a technique to improve the recovery after a frame erasure is proposed. The technique consists in a constrained excitation search at the encoder and a resynchronization procedure at the decoder. The constraint aims at reducing the contribution of the adaptive codebook by making the innovation codebook partially model the pitch excitation. Further, for highly voiced frames, the pitch-related information contained in the innovation excitation is exploited at the decoder to speed up the resynchronization of the adaptive codebook after a frame erasure. When applied to the adaptive multirate wideband (AMR-WB) codec, the method brings a significant improvement in the case of frame erasures, at the cost of a minor quality loss compared to the standard codec at the same bit rate. The method does not need additional delay and has the advantage of maintaining full interoperability between the standard codec and its modified version.

关键词： adaptive codebook code-excited linear prediction (CELP) error propagation frame erasure wideband speech coding

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