Mediumband speech encoding using time domain harmonic scaling and adaptive residual coding

Melsa, James L.; Pande, A.

doi:10.1109/icassp.1981.1171226

Cited by 6 publications

(4 citation statements)

References 5 publications

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“…straightforward time-domain algorithms proposed in [21 seem to be more suitable for a simpler implementation without renouncing to speech quality. TDHS algorithms here exploited have been successfully combined with CVSD CH 1746•718210000 0212 $ 00.75 © 1982 IEEE 212 [4], ARC [5], SBC and ATC systems [6], usually for scaling the speech spectrum of a factor of 2.…”

Section: Codec Structurementioning

confidence: 98%

A variable rate embedded-code speech waveform coder

Copperi

ICASSP '82. IEEE International Conference on Acoustics, Speech, and Signal Processing

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The design of variable rate coders for operation at 9.6 to 16 kbps that provide high speech quality and maintain robustness to environmental impairments, while retaining a low complexity, is an area of current research.An approach is here developed to achieve this objective, by exploiting a combination of Time Domain Harmonic Scaling algorithms and variable rate embedded-code ADPCM. The novel system, deeply examined and subjectively evaluated, emerges as a viable method for speech encoding, providing a quality equivalent to that of plain ADPCM at data rates of 24 to 48 kbps. INTRODUCTONVariable rate embedded-code digitizers are very attractive in a variety of speech processing applications such as packet switching, speech interpolation, voice-storage and message store-and-forward. The main feature which underlies the embedded-code structure, is that the rate change takes place on the digital channel and can be simply accomplished by deleting and inserting set of bits, without any elaborate code conversion. The actual data rate is determined by a control signal, according to the required network throughput, thus an effective management of link utilization, queue content and source activity is allowed. This paper deals with the design and performance analysis of an embedded-code Adaptive DPCM [1] combined with Time Domain Harmonic Scaling ITDHS) algorithms [2].Two main schemes are considered, with and without entropy coding, which operate in the range 9.6-16 kbps, with inputs bandlimited to 3.2 kHz and sampled at 6.4 kHz.Several important advantages emerge because of the waveform coding technique used, like high robustness to background noise and in tandem ing connection with other coders, good performance in presence of in-band non speech signals and, moreover, a low hardware complexity due to a computationally simple algorithm. System details are presented in Section 2. Section 3 discusses the experimental evaluation of the optimized codec. CODEC STRUCTUREA block diagram of the codec is shown in Fig. 1. Since ADPCM alone cannot provide acceptable quality at bit rates lower than about 16 kbps. , a rate-change algorithm is inserted as preprocessing end to halve the sampling rate of the input signal, so to enable the quantizer to use twice as many bits per sample.Even if a rigorous approach for frequency-scale modification of signals is based on the short-time Fourier analysis L3]. straightforward time-domain algorithms proposed in [21 seem to be more suitable for a simpler implementation without renouncing to speech quality. TDHS algorithms here exploited have been successfully combined with CVSD CH 1746•718210000 0212 $ 00.75 © 1982 IEEE 212[4], ARC [5], SBC and ATC systems [6], usually for scaling the speech spectrum of a factor of 2. TDHS AlgorithmsBasically, these algorithms change the speech rate by discarding or repeating short pieces of waveform which are at least equal in length to a pitch period. A refinement of a crude cut-and-splice method exploits the pitch information to perform a time-varying weightin...

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Section: Codec Structurementioning

confidence: 98%

A variable rate embedded-code speech waveform coder

Copperi

ICASSP '82. IEEE International Conference on Acoustics, Speech, and Signal Processing

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“…This value of the pitch period is transferred to the Time Domain Harmonic Compression algorithm which produces N0 samples of compressed speech y(k) as described by Melsa and Pande [1].…”

Section: System Configurationmentioning

confidence: 99%

“…However, various objective measure criteria [1] were used to evaluate the performance of DPCM coder.…”

Section: Performancementioning

confidence: 99%

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Mediumband speech encoding using time-domain harmonic scaling and adaptive residual coding for noisy channels

Melsa

Pande²

ICASSP '82. IEEE International Conference on Acoustics, Speech, and Signal Processing

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This paper describes the study of an approach to speech digitization at mediumband bit rates of 9.6 kb/s to 16 kb/s for the noisy channels.The technique is based on a combination of Time Domain Harmonic Scaling and an Adaptive Residual Coding. The use of fixed wordsize codewords and the absence of any side information transmission makes the system structure quite simple.Simulation studies have shown that this technique produces communication quality speech at 9.6 kb/s and excellent quality speech at 16 kb/s with a channel bit-error-rate as high as 1%.

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Frequency Scaling of Speech Signals by Transform Techniques

Malah

Flanagan²

1981

Bell System Technical Journal

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The general framework of short‐time Fourier analysis, modification, and synthesis is used to describe in a unified way several known techniques for frequency scaling of speech signals. Subsequently, a frequency domain harmonic scaling technique is studied in detail with emphasis on improving its performance and its implementation efficiency. This technique is particularly attractive for 2:1 scaling by use of a sign tracking algorithm which avoids the need for explicit phase computation and unwrapping. The implementation efficiency is achieved by using the fast Fourier transform algorithm, embedded decimation and interpolation, and an extended version of a recently developed weighted overlap‐add synthesis scheme. The improvement in quality is achieved by improved sign tracking and elaborate design and selection of the analysis and synthesis prototype filters (data windows). Results of computer simulations, for a variety of adverse acoustical environment conditions, indicate that the system is highly robust but its quality for clean speech is lower than with a time domain harmonic scaling technique which uses pitch information. In applications which do not permit pitch transmission, a hybrid scheme which combines the two techniques is found to yield a better quality than either system alone.

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Mediumband speech encoding using time domain harmonic scaling and adaptive residual coding

Cited by 6 publications

References 5 publications

A variable rate embedded-code speech waveform coder

A variable rate embedded-code speech waveform coder

Mediumband speech encoding using time-domain harmonic scaling and adaptive residual coding for noisy channels

Frequency Scaling of Speech Signals by Transform Techniques

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