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检索条件"主题词=Linear predictive coding"

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Use of low-delay code-excited linear prediction technology in circuit multiplexed networks

Use of low-delay code-excited linear prediction technology i...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： S. Dimolitsas F.L. Corcoran M. Baraniecki J.G. Phipps COMSAT Laboratories Clarksburg MD USA Intelsat Limited Washington D.C. DC USA

The LD-CELP (code excited linear prediction) algorithm was adopted by the CCITT, as a Recommendation G.728 for the coding of speech with toll quality at 16 kbit/s. The operation of the LD-CELP algorithm at 12.8 kbit/s is described, and its performance is assessed both with voice and nonvoice signals in single and interconnected network configurations. The 12.8 kbit/s LD-CELP codec is found to perform equivalently to 24 kbit/s ADPCM (adaptive differential pulse code modulation). It is also shown that 12.8 kbit/s LD-CELP is acceptably transparent to network signaling. As a result, it can be concluded that the operation of the LD-CELP algorithm at 12.8 kbit/s presents a viable option for inclusion within 16-kbit/s-based DCME (digital circuit multiplication equipment) or PCME (packet circuit multiplication equipment) overload strategies.< >

关键词： Intelligent networks Speech coding Telephony Filters Integrated circuit interconnections Pulse modulation linear predictive coding Speech analysis Delay Decoding

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Development of a 4.8-9.6 kbps RELP vocoder

Development of a 4.8-9.6 kbps RELP vocoder

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： M. Dankberg D. Wong Space and Secure Telecommnications Systems Division Rockwell International Newport Beach CA USA

A new version of the Residual Excited linear predictive (RELP) vocoder has been simulated. The objective has been to reduce the data rate required for good quality speech to 4.8 kbps. Results have indicated that it is possible to remove the hoarseness currently associated with low data rate RELP speech. Development of a pitch predictive ADPCM residual encoder and preliminary results on new harmonic generation techniques are discused. Taped demonstrations will be played at the conference.

关键词： Vocoders Baseband linear predictive coding Space technology Speech coding Speech enhancement Noise robustness Acoustic waves Humans Telecommunications

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Fine granularity scalability for MP-CELP based speech coding with HPDR technique

Fine granularity scalability for MP-CELP based speech coding...

引用

IEEE Asia-Pacific Conference on Circuits and Systems

作者： S. Chompun Electrical Engineering Division Faculty of Engineering at Si racha Kasetsart University Chonburi Thailand

In this work, based on the MP-CELP speech coding with HPDR technique, fine granularity scalability (FGS) is introduced by adjusting the amount of transmitted fixed excitation information. The FGS feature aim at changing the bit rate of the conventional coding more finely and more smoothly. Through performance analysis and computer simulation, the quality of scalability of the MP-CELP coding is presented with an improvement from conventional scalable MP-CELP. The HPDR technique is also applied to the MP-CELP to use for tonal language, meanwhile it can support the core coding rate of 4.2, 5.5, 7.5 kbps and additional scaled bit rates.

关键词： Scalability Speech coding Bit rate MPEG 4 Standard Natural languages linear predictive coding Speech synthesis Decoding Stochastic processes Performance analysis

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Noise tolerance improvement of the linear Periodic Model reference adaptive control using vigorous probing

Noise tolerance improvement of the Linear Periodic Model ref...

引用

IEEE International Conference on Electro-Information Technology

作者： Naghmeh Mansouri Madjid Fathi Department of Electrical and Computer Engineering University of Waterloo Waterloo Canada Department of Electrical and Computer Engineering University of Siegen Siegen Germany

The use of a linear periodic controller (LPC) has been proposed as a new approach in the field of model reference adaptive control. The resulting controller can handle rapid changes in plant parameters, and it provides smooth transient behavior for a closed-loop system. Moreover, the LPC generates control signals, which are modes in size when measured using the infinity norm. Although the LPC has these advantages, it suffers from poor noise tolerance. The smaller the sampling time is the less noise tolerant the controller is. In this work, to alleviate this drawback, we apply a probing signal with a larger size. The probing size is inversely proportional to the sampling time. The proposed method has significantly better noise rejection but larger control signal.

关键词： Adaptive control linear predictive coding Sampling methods Control systems Size control Parameter estimation Computer science Signal generators Size measurement H infinity control

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A hybrid input/output spectrum adaptation scheme for LD-CELP coding of speech

A hybrid input/output spectrum adaptation scheme for LD-CELP...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Siu-Pun Chui Cheung-Fat Chan Department of Electronic Engineering City University of Hong Kong Hong Kong China

ISBN: (纸本)0780331923

This paper describes a hybrid input/output block adaptation scheme for the quantization of spectral parameters for low-delay CELP coders. In this scheme, line spectrum pair (LSP) coefficients are used as parameters for spectral adaptation and they are partitioned into two segments; LSPs in the first segment which corresponding to low frequency regions are backward adapted from the reproduction speech while LSPs in the second segment are adapted from input speech and quantized as side information to the decoder. The rationale behind this scheme is to compensate for the spectrum distortion introduced by conventional backward adaptation schemes operating at low bit rates. Simulation results show that the hybrid LSP adaptation scheme achieves transparent quantization at 12 bits for each analysis frame of 5 ms.

关键词： Speech coding Delay Quantization Bit rate Speech analysis Acoustic noise Speech processing Speech enhancement linear predictive coding Frequency

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Least squares-adapted edge-look-ahead prediction with run-length encodings for lossless compression of images

Least squares-adapted edge-look-ahead prediction with run-le...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Lih-Jen Kau Yuan-Pei Lin Department Elec. and Control Engineering National Chiao Tung University Hsinchu Taiwan

Many coding methods are more efficient with certain types of images than others. In particular, run-length coding is very useful for coding areas of little changes. Adaptive predictive coding achieves high coding efficiency for fast changing areas like edges. In this paper, we propose a switching coding scheme that will combine the advantages of both run-length and adaptive linear predictive coding (RALP) for lossless compression of images. For pixels in slowly varying areas, run-length coding is used; otherwise LS (least square)-adapted predictive coding is used. Instead of performing LS adaptation in a pixel-by-pixel manner, we adapt the predictor coefficients only when an edge is detected so that the computational complexity can be significantly reduced. For this, we propose an edge detector using only causal pixels. This way, the predictor can look ahead if the coding pixel is around an edge and initiate the LS adaptation in advance to prevent the occurrence of a large prediction error. With the proposed switching structure, very good prediction results can be obtained in both slowly varying areas and pixels around boundaries as we will see in the experiments.

关键词： Image coding Image edge detection Computational complexity predictive coding Encoding linear predictive coding Detectors Bit rate Biomedical imaging Remote sensing

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Interframe differential vector coding of line spectrum frequencies

Interframe differential vector coding of line spectrum frequ...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： E. Erzin A.E. Cetin Department of Electrical and Electronics Engineering Bilkent University Ankara Turkey

In many vocoders LSFs (line spectrum frequencies) are used to encode the linear predictive coding (LPC) parameters. An interframe differential coding scheme is presented for LSFs. The LSFs of the current speech frame are predicted by using both the LSFs of the previous frame and some of the LSFs of the current frame. Then the difference vector resulting from prediction is vector quantized. The proposed scheme is computationally efficient and easy to implement, and can be used in low-bit-rate vocoders.< >

关键词： Frequency linear predictive coding Speech coding Vectors Nonlinear filters Vocoders Speech synthesis Polynomials predictive models Signal synthesis

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Network echo canceller based on the practical adaptive filter

Network echo canceller based on the practical adaptive filte...

引用

IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： Jae Ha Yoo Dong Yon Kim Department of Electronic Engineering Hankyong National University South Korea

In this paper, a new network echo canceller based on the practical adaptive filter is proposed. The proposed adaptive filter practically modifies the lattice transversal joint (LTJ) adaptive filter. It takes advantage of information in the speech decoder and coefficients of the transversal filter part in the LTJ adaptive filter are updated every other sample instead of every sample. Total complexity of the proposed adaptive filter is lower than that of the transversal filter. And the residual echo signal is decreased by residual echo cancellation using the lattice predictor whose order is less than 10. Computational complexity of the proposed echo canceller is lower than that of the transversal filter but the convergence speed is faster than that of the transversal filter. The performance of the proposed network echo canceller was verified by the experiments using the real speech signal.

关键词： Echo cancellers Adaptive filters Transversal filters Lattices Computational complexity Convergence linear predictive coding Decoding Speech processing Natural languages

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Low bit-rate speech coder based on a long-term model

Low bit-rate speech coder based on a long-term model

引用

IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： O. Lev D. Malah Dept. of Electr. Eng. Technion-Israel Inst. of Technol. Haifa Israel Department of Electrical Engineering Technion Israel Institute of Technology Israel

We present a low bit rate speech coder based on a long-term model (LTM) for voiced speech, and on the WI coder. In the LTM, a periodic input signal undergoes a time-varying spectral shaping representing the evolution of the pitch-cycle waveform. The resulting signal, which has a fixed pitch period but a time-varying pitch-cycle waveform, is multiplied by a time-varying gain function that represents the variation in speech loudness. The resulting signal then undergoes a time-axis warping, which represents the evolution of the pitch period, yielding the output speech signal. The spectral shaping in the proposed coder is based on WI. In WI, speech (or LPC residual) is observed as a continuously evolving sequence of pitch cycle waveforms. A subset of these waveforms is extracted and coded. In the decoder, after inverse quantization, missing waveforms are synthesized by interpolation. The extracted waveforms are normalized to a fixed length and sequentially aligned using a cyclical shift. Then, a two-dimensional surface, called prototype waveform surface or characteristic waveform (CW) is produced from these waveforms.

关键词： Speech analysis Speech synthesis Surface waves Interpolation linear predictive coding Decoding Energy states Gaussian noise Signal analysis Bit rate

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Code-excited linear prediction(CELP): High-quality speech at very low bit rates

Code-excited linear prediction(CELP): High-quality speech at...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： M. Schroeder B. Atal Drittes Physikalisches Institut University of Goettingen Germany AT and T Bell Laboratories Inc. Murray Hill NJ USA

We describe in this paper a code-excited linear predictive coder in which the optimum innovation sequence is selected from a code book of stored sequences to optimize a given fidelity criterion. Each sample of the innovation sequence is filtered sequentially through two time-varying linear recursive filters, one with a long-delay (related to pitch period) predictor in the feedback loop and the other with a short-delay predictor (related to spectral envelope) in the feedback loop. We code speech, sampled at 8 kHz, in blocks of 5-msec duration. Each block consisting of 40 samples is produced from one of 1024 possible innovation sequences. The bit rate for the innovation sequence is thus 1/4 bit per sample. We compare in this paper several different random and deterministic code books for their effectiveness in providing the optimum innovation sequence in each block. Our results indicate that a random code book has a slight speech quality advantage at low bit rates. Examples of speech produced by the above method will be played at the conference.

关键词： Bit rate Technological innovation Speech coding Books Feedback loop Nonlinear filters Speech synthesis predictive models linear predictive coding predictive maintenance

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