The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for cellular communications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. We feel that this approach will not only point out key references but will also provide valuable background to the beginner. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. We then proceed with descriptions of waveform coders, sinusoidal transform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. Finally, we present concluding remarks followed by a discussion of opportunities for future research. >

Speech coding: a tutorial review

A personal communications system is described which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows® environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal processors, and hardware to enable voice, fax and data communication with a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware component and from the software components operating within the personal computer. The major functions of the system are a telephone function, a voice mail function, a fax manager function, a multi-media mail function, a show and tell function, a terminal function and an address book function. The telephone function allows the present system to operate, from the users perspective, as a conventional telephone using either hands-free, headset or handset operation. The telephone function is more sophisticated than a standard telephone in that the present system converts the voice into a digital signal which can be processed with echo cancellation, compressed, stored as digital data for later retrieval and transmitted as digital voice data concurrent with the transfer of digital information data.

Computer-based multifunction personal communications system.

Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel. >

Advances in speech and audio compression

A low-delay code-excited linear prediction (LD-CELP) speech coder which is expected to be standardized in 1992 as a CCITT G Series Recommendation for universal applications of speech coding at 16 kb/s is presented. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive and by using a small excitation vector size of five samples. The official CCITT laboratory tests revealed that the speech quality of this 16 kb/s LD-CELP coder is either equivalent to or better than that of the CCITT G.721 standard 32-kb/s ADPCM coder for almost all conditions tested. A description of the LD-CELP algorithm, its implementation on the DSP32C for CCITT testing, and performance results from these tests are presented. >

A low-delay CELP coder for the CCITT 16 kb/s speech coding standard

A low-bitrate (typically 8 kbit/s or less), low-delay digital coder and decoder based on Code Excited Linear Prediction for speech and similar signals features backward adaptive adjustment for codebook gain and short-term synthesis filter parameters and forward adaptive adjustment of long-term (pitch) synthesis filter parameters. A highly efficient, low delay pitch parameter derivation and quantization permits overall delay which is a fraction of prior coding delays for equivalent speech quality at low bitrates.

Method and apparatus for low-delay celp speech coding and decoding

A method of encoding speech sounds to facilitate their transmission to and reconstruction at a remote receiver. A transmitter and a receiver have identical filters and identical codebooks containing prestored excitation vectors which model quantized speech sound vectors. The speech sound vectors are compared with filtered versions of the codebook vectors. The filtered vector closest to each speech sound vector is selected. During the comparison, filtration parameters derived by backward predictive analysis of a series of previously selected filtered codebook vectors are applied to the filter. The transmitter sends the receiver an index representative of the location of the selected vector within the codebook. The receiver uses the index to recover the selected vector from its codebook, and passes the recovered vector through its filter to yield an output signal which reproduces the original speech sound sample. By applying the same backward predictive analysis technique employed by the transmitter to the same series of previously selected filtered codebook vectors to which the transmitter applied the technique, the receiver derives the same combination of filtration parameters which the transmitter applied to its filter while selecting the codebook vector corresponding to the transmitted index.

Low-delay vector backward predictive coding of speech

Many of the advances in speech coding in the past decade at rates of 48–16 kbit/s have been based on excitation coding by means of analysis-by-synthesis Excitation coding schemes have a decoder structure consisting of an excitation signal applied to a time-varying synthesis filter to produce the reconstructed, or “synthesized,” output speech In addition to other tasks, the encoder must determine a suitable excitation signal and transmit data that specifies this excitation In the analysis-by-synthesis technique, the excitation is selected by a closed-loop search procedure where a candidate excitation signal segment is applied to the synthesis filter, the synthesized waveform is compared with the original speech segment, the distortion is measured, and the process is repeated for all excitation segments stored in an excitation codebook The index of the “best” excitation segment is transmitted to the decoder, which retrieves the excitation segment from a codebook identical to that at the encoder The parameters of the synthesis filter are computed using well-known linear prediction analysis techniques on a frame of buffered input samples and transmitted to the decoder This coding scheme is often called Vector Excitation Coding (VXC) or Code Excited Linear Prediction (CELP) [1,2]

Backward Adaptive Configurations for Low-Delay Vector Excitation Coding

To attain a very-low-delay speech coder at 16 kb/s while maintaining a quality acceptable for the public switched telephone network, low delay vector excitation coding (LD-VXC) is introduced. Backward adaptation is used to track the spectral characteristics of the signal without requiring any buffering of the input speech, thereby allowing a very low delay to be achieved in an analysis-by-synthesis structure. The algorithm differs markedly from conventional VXC or CELP (code-excited linear prediction) coders due to the use of backward adaptive linear prediction for modeling the time-varying short- and long-term correlation of speech. The LD-VXC coder provides very good speech quality at 16 kb/s, moderate complexity, a delay of under 2 ms, and a gentle degradation of quality with transmission errors. The algorithm was submitted to the CCITT as a candidate for a future 16-kb/s speech coding standard. >

Backward adaptation for low delay vector excitation coding of speech at 16 kbit/s

The need for a 16-kb/s speech coding algorithm that has very low coding delay while achieving essentially the same high quality as the 32-kb/s adaptive differential pulse code modulation (ADPCM) standard G.721 is addressed. The authors describe low-delay vector excitation coding (LD-VXC), a new coding algorithm which provides high quality with less than 2 ms of coding delay and is robust to transmission errors. The algorithm combines techniques such as vector quantization, analysis-by-synthesis, and perceptual weighting together with backward adaptive linear predictive encoding, and uses a novel long-term predictor employing backward adaptive pitch tracking. Perceptually based nose shaping and postfiltering contribute to the masking of audible quantization noise. >

Low-delay vector excitation coding of speech at 16 kb/s

A pitch predictor is used in speech coding systems to remove distant-sample redundancies in speech signals due to the periodicity of voiced segments of speech. This paper describes a backward-adaptive pitch prediction algorithm which operates in conjunction with a backward-adaptive short-term predictor in a low-delay speech coding system operating at 16 kbit/s. Backward prediction in the analysis-by-synthesis environment has been introduced in [1]. The complete low-delay speech codec for which this pitch predictor was designed, Low-Delay Vector Excitation Coding (LD-VXC), will be presented in the companion paper [2]. Backward pitch prediction was been previously mentioned in the literature [3], but in a different environment than what will be discussed in this paper.

Robert Pettigrew

Papers

Low-delay vector backward predictive coding of speech

Backward Adaptive Configurations for Low-Delay Vector Excitation Coding

Backward adaptation for low delay vector excitation coding of speech at 16 kbit/s

Low-delay vector excitation coding of speech at 16 kb/s

Hybrid Backward Adaptive Pitch Prediction for Low-Delay Vector Excitation Coding