The recently standardized codec for Enhanced Voice Services (EVS) consists of a number of modes to achieve its high coding flexibility. In this paper we focus on techniques that enable a seamless switching between two linear prediction based modes running at different sampling rates within this codec. The first one deals with an efficient conversion of the linear prediction filter coefficients. The other one is based on a constrained-memory ACELP called transition coding (TC) that significantly limits the inter-frame long-term dependency. We show that the use of TC can be successfully extended to improve quality also in coding other transitions, e.g. strong onsets of voiced speech.

In this paper a novel technique is presented to efficiently mix traditional ACELP time domain coding with a frequency domain coding model to improve the quality of generic audio signals coded at low bitrates without additional delay. The paper discusses how to integrate parts of a traditional Algebraic Code Excited Linear Prediction (ACELP) speech codec to create a time-domain contribution which coexists with a frequency based coding model. A mechanism to determine the value of the time-domain contribution is proposed and a method is described how the frequency-domain contribution might be added without increasing the overall delay of the codec. The proposed method forms part of the recently standardised 3GPP EVS codec.

This paper presents two new post-processing techniques to address limitations of the deployed low bit rate speech codecs in case of unvoiced speech and background noise, and in case of music. Both post-processing techniques enhance the spectrum of the decoded excitation signal without increasing the codec algorithmic delay. The paper discusses how to integrate the enhancement procedure of unvoiced speech and background noise and of generic audio signals coded by low bit rate ACELP codecs. The proposed post-processing procedures are part of the AMR-WB interoperable modes of the recently standardized 3GPP EVS codec [1].

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding.

In this paper we present a novel technique to enhance music signals encoded using a low bit rate CELP coder. The method is based on reduction of inter-tone quantization noise for decoded music signals without affecting the quality for speech signals. The proposed technique consists of two modules. The first module is used to discriminate between stable tonal sounds and other sounds and the second module is used to reduce the inter-tone quantization noise in the stable tonal segments. The inter-tone noise is reduced by means of spectral subtraction. The proposed method is a part of the newly standardised ITU-T G.718 codec.

This paper describes the scalable coder - G.729.1 - which has been recently standardized by ITU-T for wideband telephony and voice over IP (VoIP) applications. G.729.1 can operate at 12 different bit rates from 32 down to 8 kbit/s with wideband quality starting at 14 kbit/s. This coder is a bitstream interoperable extension of ITU-T G.729 based on three embedded stages: narrowband cascaded CELP coding at 8 and 12 kbit/s, time-domain bandwidth extension (TDBWE) at 14 kbit/s, and split-band MDCT coding with spherical vector quantization (VQ) and pre-echo reduction from 16 to 32 kbit/s. Side information - consisting of signal class, phase, and energy - is transmitted at 12, 14 and 16 kbit/s to improve the resilience and recovery of the decoder in case of frame erasures. The quality, delay, and complexity of G.729.1 are summarized based on ITU-T results.

Error propagation after a frame loss is an important factor in quality degradation for predictive speech coders. This is mainly due to the lack of synchronization in the adaptive codebook in the good frames following a frame erasure. This article presents a method for resynchronizing the glottal pulse after an erased frame. The method uses an extra frame delay at the decoder, and can be applied with or without additional side information. The approach has been implemented in the G.729.1 standard and results in improved decoder convergence after erased frames. Subjective tests have demonstrated that this improves perceived quality in the presence of frame erasures.

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

This paper presents a hybrid audio coding algorithm integrating an LP-based coding technique and a more general transform coding technique. ACELP is used in LP-based coding mode, whereas algebraic TCX is used in transform coding mode. The algorithm extends previously published work on ACELP/TCX coding in several ways. The frame length is increased to 80 ms, adaptive multi-length sub-frames are used with overlapping windowing, an extended multi-rate algebraic VQ is applied to the TCX spectrum to avoid quantizer saturation, and noise shaping is improved. Results show that the proposed hybrid coder has consistently high performance for both speech and music signals.

Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase, 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides a unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

The description and design of the source-controlled variable-rate multimode wideband (VMR-WB) codec recently selected by the 3/sup rd/ Generation Partnership Project 2 (3GPP2) for the cdma2000/spl reg/ system in Rate-Set II are presented. The paper gives an overview of the codec and the methodologies that enable high quality wideband coding at average data rates ranging from TIA/EIA/IS-733 ADR (average data rate) to that of TIA/EIA/IS-127. The codec has three modes of operation at different average data rates and a fourth mode that is interoperable with 3GPP/AMR-WB (ITU-T G.722.2). Despite the interoperability constraint, the codec is capable of meeting the aggressive performance requirements through the use of novel techniques such as noise suppression, efficient signal classification, new coding types optimized for stable voiced and unvoiced frames, novel post-processing technique for periodicity enhancement in the lower frequency band, and improved frame erasure concealment mechanisms.

We present an objective and subjective comparison of alternative methods for improving the robustness of speech coders in packet networks. The two approaches are considered: 1) adding redundancy in the packets to improve the robustness of a baseline encoder; 2) reducing (or eliminating) inter-frame dependencies at the encoder. It is shown that both approaches have to trade bit rate and/or delay for quality over lossy channels. Formal subjective tests clearly show that, using relatively simple forward error correction methods, standard coders such as ITU-T recommendation G.729 can be made significantly more robust than "frame-independent" coders, at a lower or similar bit rate.

This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

We present in this paper new methods for achieving high- quality wideband speech at low rates using the ACELP algorithm. Several innovations are introduced to optimize the quality and minimize the complexity of the coder. A multi-rate wideband speech encoding algorithm based on these techniques was recently selected by 3GPP as the standard for AMR-WB, and is currently one of the candidates for the ITU- T wideband speech coder standard at around 16 kbit/sec. This standard was jointly developed by VoiceAge and Nokia.

The Third Generation Partnership Project (3GPP) and European Telecommunications Standards Institute (ETSI) have carried out development and standardisation of a wideband speech codec for GSM and the third generation mobile communication WCDMA system since 1999. The Adaptive Multi-Rate Wideband (AMR-WB) codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. The AMR-WB codec was jointly developed by Nokia and VoiceAge. AMRWB extends the audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to existing 2nd and 3rd generation mobile communication systems. The wideband speech service provided by the AMRWB codec will give mobile communication speech quality that even exceeds (narrowband) wireline quality.

This paper describes a fixed-rate quantizer using algebraic techniques as well as conventional stored codebooks to represent the wideband line spectral frequencies (LSF). It is based on two-stage split vector quantization with random and lattice codebooks in the first and second stage, respectively. We investigate the use of variable bit allocation in lattice quantization in order to capture outliers and reduce the overload distortion in spectral quantization, particularly for non-speech signals. Experimental results show that a variable bit allocation in split quantization yields performances slightly better than a fixed allocation at the same rate.

This paper describes the 8 kb/s speech coding algorithm G.729 which has been standardized by ITU-T. The algorithm is based on a conjugate-structure algebraic CELP (CS-ACELP) coding technique and uses 10 ms speech frames. The codec delivers toll-quality speech (equivalent to 32 kb/s ADPCM) for most operating conditions. This paper describes the coder structure in detail and discusses the reasons behind certain design choices. A 16-b fixed-point version has been developed as part of Recommendation G.729 and a summary of the subjective test results based on a real-time implementation of this version are presented.

In this paper, we describe the enhanced full rate (EFR) speech codec that has recently been standardised for the North American TDMA digital cellular system (IS-136). The EFR codec, specified in the IS-641 standard, has been jointly developed by Nokia and University of Sherbrooke. The codec consists of 7.4 kbit/s speech (source) coding and 5.6 kbit/s channel coding (error protection) resulting in a 13.0 kbit/s gross bit-rate in the channel. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec offers speech quality close to that of wireline telephony (G.726 32 kbit/s ADPCM used as a wireline reference) and provides a substantial improvement over the quality of the current speech channel. The improved speech quality is not only achieved in error-free conditions, but also in typical cellular operating conditions including transmission errors, environmental noise, and tandeming of speech codecs.

This paper describes the GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system. The GSM EFR codec has been jointly developed by Nokia and University of Sherbrooke. It provides speech quality at least equivalent to that of a wireline telephony reference (32 kbit/s ADPCM). The EFR codec uses 12.2 kbit/s for speech coding and 10.6 kbit/s for error protection. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec provides substantial quality improvement compared to the existing GSM full rate and half rate codecs. The old GSM codecs lack wireline quality even in error-free channel conditions, while the EFR codec provides wireline quality not only for error-free conditions but also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile to mobile calls).

This paper describes the recently adopted ITU-T Recommendation G.729 Annex A (G.729A) for encoding speech signals at 8 kbit/s with low complexity. G.729A has been selected as the standard speech coding algorithm for multimedia digital simultaneous voice and data (DSVD). G.729A is bitstream interoperable with G.729; i.e., speech coded with G.729A can be decoded with G.729, and vice versa. As G.729, it uses the conjugate structure algebraic code excited linear prediction (CS-ACELP) algorithm with 10 ms frames. However, several algorithmic changes have been introduced into G.729 which resulted in 50% drop in its complexity, enabling a DSP implementation with a complexity of about 10-12 MIPS. This paper describes the algorithmic changes which have been introduced in order to achieve the low complexity goal while meeting the terms of reference. Subjective tests have been performed by ITU-T in both the selection phase and the characterization phase and the results showed that the performance of G.729A is equivalent to both G.729 and G.726 at 32 kbit/s in most operating conditions; however, it is slightly worse in case of three tandems and in the presence of background noise. A breakdown of the complexities of both G.729 and G.729A is given at the end of the paper.

This paper describes the enhanced full rate (EFR) speech codec that has been standardized by ETSI for the GSM mobile communications system in 1996. The codec was developed jointly by Nokia and the University of Sherbrooke. It operates at 12.2 kbit/s speech coding (source coding) bit-rate and provides speech quality equivalent to that of wireline telephony (G.726 32 kbit/s ADPCM). The algorithm is based on the algebraic code-excited linear prediction (ACELP) technology, using 20 ms speech frames. The GSM EFR speech codec provides substantial quality improvement compared to the current GSM full rate (FR) and half rate (HR) codecs. The old GSM codecs lag far behind wireline quality even in error-free conditions, while the EFR codec provides wireline quality also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile-to-mobile calls). The codec was defined using fixed-point basic operators with complexity estimated at 18 WMOPS (below that of the GSM half-rate codec).

This paper presents an algorithm for encoding speech signals at bit rates below 4 kbit/s based on a mixed harmonic and stochastic modeling of the excitation signal. The algorithm uses robust pitch tracking and efficient voicing analysis to determine the ratios of the harmonic and stochastic components. The harmonic component is synthesized using a bank of bandpass filters while the stochastic component is synthesized using inverse STFT with overlap-and-add. Postfiltering is utilized at the decoder to enhance the quality of synthesized speech. A 2.4 kbit/s version of the algorithm was formally tested and the DAM and DRT scores showed that the coder performance is comparable to that of DoD 4.8 kbit/s Federal Standard FS-1016.

This paper describes the application of transform coded excitation (TCX) coding to encoding wideband speech and audio signals in the bit rate range of 16 kbits/s to 32 kbits/s. The approach uses a combination of time domain (linear prediction; pitch prediction) and frequency domain (transform coding; dynamic bit allocation) techniques, and utilizes a synthesis model similar to that of linear prediction coders such as CELP. However, at the encoder, the high complexity analysis-by-synthesis technique is bypassed by directly quantizing the so-called target signal in the frequency domain. The innovative excitation is derived at the decoder by inverse filtering the quantized target signal. The algorithm is intended for applications whereby a large number of bits is available for the innovative excitation. The TCX algorithm is utilized to encode wideband speech and audio signals with a 50-7000 Hz bandwidth. Novel quantization procedures including inter-frame prediction in the frequency domain are proposed to encode the target signal. The proposed algorithm achieves very high quality for speech at 16 kbits/s, and for music at 24 kbits/s.>

A toll quality speech codec at 8 kbit/s with a 10 ms speech-frame currently under standardization by the CCITT is presented. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP). Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Initial subjective tests showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kbit/s in error-free conditions and it outperforms G.726 under error conditions. The codec can support a frame erasure rate up to 3% with slight degradation and performs adequately under tandeming conditions. The algorithm has been implemented on a single fixed-point DSP for the CCITT qualification test. It requires about 24 MIPS.>

The application of algebraic code excited linear prediction (ACELP) coding to wideband speech is presented. An algebraic codebook with a 20 bit address can be used without any storage requirements and, more importantly, with a very efficient search procedure which allows for real-time implementation. The authors describe an efficient procedure for searching such a large codebook deploying a focused search strategy, where less than 0.1% of the codebook is searched with performance very close to that of a full search. High-quality speech at a bit rate of 13 kbps was obtained.>

A modified regular-pulse excitation (RPE) linear predictive coding (LPC) codec is used to encode audio signals of 15 kHz bandwidth. With appropriate perceptual masking and carefully selected excitation structure, a fixed-pulse excitation (FPE) LPC codec is able to provide high-fidelity audio output. The quantization of both the LPC parameters and the excitation pulse sequence is examined. It is found that the LPC parameters can be quantized efficiently by adaptive differential PCM, while the excitation pulse sequences need more bits to quantize the amplitudes compared to narrowband speech encoding. Informal listening tests show that high-fidelity audio can be achieved at bit rates less than 100 kb/s.>

The performance of error protected transformed binary pulse excited (TBPE) coders over Rayleigh fading channels is investigated. The TBPE coder is a stochastically excited linear predictive coding (LPC) coder which produces near-toll quality speech in the vicinity of 8 kb/s utilizing an extremely simple excitation search procedure. A TBPE coder operating at 7.4 kb/s is used, and two methods of embedding channel coding into the speech coder are studied, aiming at the overall bit rate of 11.4 kb/s specified for the half-rate GSM coder. A method of embedding Reed-Solomon codes into speech coding is presented, and it is compared with rate-compatible punctured convolutional (RCPC) codes. Results show the superiority of the proposed scheme over RCPC codes in both performance and implementation simplicity.>

A computationally efficient stochastic vector quantization (VQ) scheme for the quantization of LPC parameters with less than 25 bits is proposed. The complexity of the self-excited vocoder (SEV) is reduced by using fast algorithms based on an efficient representation of the innovation sequence. When the suggested VQ scheme is deployed in the improved SEV, good communications speech quality is obtained at 4.8 kb/s. When the bit rate is increased to 6.8 kb/s, near-toll-quality speech is achieved.>