In the context of signal reconstruction and coding, a new robust parametric formulation to linear predictive coding (LPC) is introduced. The linear prediction filter coefficients are transformed into a set of weighted...
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In the context of signal reconstruction and coding, a new robust parametric formulation to linear predictive coding (LPC) is introduced. The linear prediction filter coefficients are transformed into a set of weighted line frequencies. The positive weights play the dual role of a new set of parameters and simultaneously they exhibit the relative importance of the associated line frequencies. This new representation for LPC is shown to be always stable under quantization.
The STU-III program has been very successful in providing secure, high quality communications. The STU-III program, however, has so far been restricted to strategic networks. Tactical networks use devices such as SINC...
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The STU-III program has been very successful in providing secure, high quality communications. The STU-III program, however, has so far been restricted to strategic networks. Tactical networks use devices such as SINCGARS, VINSON, and MINTERM (tactical terminal) which can not interoperate with the STU-III network. The paper discusses the modifications to MINTERM that provide it with a STU-III-compatible mode. It is shown that most of the changes can be made in software and that the modified MINTERM can interoperate with the STU-III network over a large variety of media.
A new technique for the determination of the LPC coefficients is introduced. In this method the LPC coefficients are shown to be the weighting factors of a simple neuron. This makes this technique suitable for a lot o...
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A new technique for the determination of the LPC coefficients is introduced. In this method the LPC coefficients are shown to be the weighting factors of a simple neuron. This makes this technique suitable for a lot of applications that uses LPC as a signal processing tool like speech processing and production. The error introduced by this method is comparable to the error obtained by the conventional methods like the autocorrelation method.
The paper describes a comparison of a C implementation of a linearpredictive voice coder (LPC) and an implementation based on Spectron Microsystem's Signal Processing Operating System (SPOX). The hardware platfor...
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The paper describes a comparison of a C implementation of a linearpredictive voice coder (LPC) and an implementation based on Spectron Microsystem's Signal Processing Operating System (SPOX). The hardware platform was a Texas Instruments TMS320C30 Evaluation Module. The SPOX and C implementations were compared based on execution time, ease of program development and maintenance, and portability to different hardware platforms. The vocoder algorithms and the results of the comparison of both implementations are presented.
A problem with speech coders based on trained quantizers is their lack of robustness against variations in the microphone and input filter response. In this paper a simple backward adaptive prefiltering technique is p...
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A problem with speech coders based on trained quantizers is their lack of robustness against variations in the microphone and input filter response. In this paper a simple backward adaptive prefiltering technique is proposed as a means of improving the robustness and quality of a speech coder at no cost in bit rate. The technique is particularly useful in conjunction with vector quantization (VQ) of the linear predictive coding (LPC) parameters. The performance of the prefilter, denoted a microphone and speaker adaptation (MSA) filter, has been evaluated in terms of prediction gain and spectral distortion, together with objective and subjective quality of a 7.5 kbit/s CELP speech coder. In this coder a 10-bit direct VQ of the LPC parameters using the residual energy distortion measure has been applied. This is consistent with the covariance method of LPC analysis. Simulation results illustrate that the MSA filter significantly improves the performance and robustness of the LPC VQ against changes in the input response. The 7.5 kbit/s CELP with a trained excitation codebook and MSA was found to be clearly better (subjectively and objectively) than the one without MSA. The coder with MSA also showed to be practically indistinguishable from the same CELP with unquantized LPC coefficients and a stochastic excitation codebook.
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...
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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.
Trellis coded vector quantization (TCVQ) and code-excited linear prediction (CELP) coding are combined to form an efficient low-bit rate speech coding system. The resulting system uses a trellis search to select the s...
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Trellis coded vector quantization (TCVQ) and code-excited linear prediction (CELP) coding are combined to form an efficient low-bit rate speech coding system. The resulting system uses a trellis search to select the synthesis filter excitation sequence, and is referred to as trellis excitation coding (TEC). Simulations are performed for encoding rates of 6.4 and 8 kbps. Informal listening tests indicate that the 8 kbps TEC system has quality roughly between that of 6-bit and 7-bit mu-law PCM with mu = 255. The 6.4 kbps TEC system provides speech quality between 5-bit and 6-bit mu-law PCM. A subjective comparison with vector sum excited linear prediction (VSELP) indicates that the 8 kbps TEC and the VSELP reconstructed speech is about equally preferable.
This paper describes a new artificial speech signal (ASVQ: Artificial Speech by Vector Quantization technique) which reflects the average characteristics of the human voice. The ASVQ is intended for use as a test sign...
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This paper describes a new artificial speech signal (ASVQ: Artificial Speech by Vector Quantization technique) which reflects the average characteristics of the human voice. The ASVQ is intended for use as a test signal in the objective evaluation of speech coding system quality. To obtain the average characteristics, a very large speech data base is analyzed. The ASVQ generation method which reflects the extracted average characteristics of the human voice is formulated. This method applies vector quantizing analysis to the speech data base. The LPC speech synthesis circuit is used to reproduce the average characteristics. Finally, the new artificial speech signal is compared with a human voice and the estimation accuracy of the subjective quality of speech coding systems and nonlinear distortions is evaluated.
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