A variable-bit-rate modulo PCM (VBR-MPCM) scheme is proposed. The modulo amplitude and the quantization bits are varied adaptively to prevent anomaly errors and increase data-compression efficiency. It is confirmed by...
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A variable-bit-rate modulo PCM (VBR-MPCM) scheme is proposed. The modulo amplitude and the quantization bits are varied adaptively to prevent anomaly errors and increase data-compression efficiency. It is confirmed by computer simulation that the performance of the VBR-MPCM scheme is about 5 dB to about 15 dB better than those of the conventional MPCM-AQF, ADPCM, and DPCM-AQF schemes.
A 16-kb/s speech codec that uses commercially available digital signal processors, and its performance, are described. The coding algorithm used is adaptive differential PCM with multiquantization. Reconstructed speec...
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A 16-kb/s speech codec that uses commercially available digital signal processors, and its performance, are described. The coding algorithm used is adaptive differential PCM with multiquantization. Reconstructed speech with a 21-dB segmental SNR was obtained. With error correction, this codec provides good speech quality even with a bit-error rate of 10**-**2-10**-**3. To improve the subjective quality of the reconstructed speech, adaptive postfiltering was also applied. Since the processing delay of this codec is less than 10 ns, no echo-canceller is needed. Moreover, 2400-b/s voice-band data could be transmitted with a data error rate from 10**-**7 to 5 multiplied by 10**-**6, and G. III facsimiles were successfully transmitted using this codec.
The digital noise power due to transmission errors is determined for 8-bit mu -law PCM ( mu equals 255) signals transmitted via M-level QAM, M equals 16, 64, and 256, over Gaussian and Rayleigh fading channels. The an...
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The digital noise power due to transmission errors is determined for 8-bit mu -law PCM ( mu equals 255) signals transmitted via M-level QAM, M equals 16, 64, and 256, over Gaussian and Rayleigh fading channels. The analytical results apply for various laws that map the 8-bit mu -law PCM signal into the QAM format and for different binary Gray codes used to represent the QAM signal points. In addition to the theoretical results expressed as overall speech signal-to-noise ratio (SNR) as a function of channel SNR for different input speech signals levels, simulations for ideal Gaussian and fading radio channels using speech signals are presented. It is concluded that the 256-level QAM system using the best mapping law and Gray code has a gain of 6 dB in overall speech SNR compared to transmission without coordination of the speech bits over a Gray coded link. Corresponding gains for 16-level and 64-level QAM are marginally smaller.
Due to the feedback architecture of the standard CCITT and ANSI 32-kb/s ADPCM (adaptive differential pulse-codemodulation) coders, large transitional errors occur when characteristics of the input signal suddenly cha...
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Due to the feedback architecture of the standard CCITT and ANSI 32-kb/s ADPCM (adaptive differential pulse-codemodulation) coders, large transitional errors occur when characteristics of the input signal suddenly change. This particularly holds for FSK modems that transmit a single-frequency signal between actual characters. A mechanism that identifies these transitions and reduces errors is presented.
A modulo-PCM with multiquantizer that has high data compression efficiency is proposed. The proposed scheme can get rid of the anomaly errors by utilizing the multiquantizer method. By using this proposed scheme, its ...
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A modulo-PCM with multiquantizer that has high data compression efficiency is proposed. The proposed scheme can get rid of the anomaly errors by utilizing the multiquantizer method. By using this proposed scheme, its performance is improved by about 10-15 dB (1. 5-1. 8 b/sample) in signal quantized noise reduction compared with the conventional scheme.
This paper describes the DTC-60A, a new 60-channel ADPCM transcoder for European CEPT networks. The DTC-60A implements a conversion between two 2048 kbps PCM streams and a single 2048 kbps ADPCM stream, using CCITT Re...
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This paper describes the DTC-60A, a new 60-channel ADPCM transcoder for European CEPT networks. The DTC-60A implements a conversion between two 2048 kbps PCM streams and a single 2048 kbps ADPCM stream, using CCITT Recommendation 32 kbps ADPCM algorithm (G. 721). NEC developed ADPCM codec LSIs are used in per channel basis for mutual conversion between PCM and ADPCM signals, thereby reducing equipment cost, size and power consumption. Other important characteristics of the DTC-60A include easy maintainability, compliance with the CCITT Recommendation (G. 761), and versatile programming and alarm display functions provided by the portable control terminal (PCT).
This paper describes test results for a set of experimental procedures designed to objectively measure the performance of four adaptive differential pulse-codemodulation (ADPCM) algorithms that use signals other than...
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This paper describes test results for a set of experimental procedures designed to objectively measure the performance of four adaptive differential pulse-codemodulation (ADPCM) algorithms that use signals other than voice. These algorithms have been under consideration by the International Telegraph and Telephone Consultative Committee (CCITT) as possible low-bit-rate coders for incorporation in digital circuit multiplication equipment. The results presented pertain to voiceband data modems operating at 9,600 bit/s or higher, and include analog performance measurements for each of the algorithms considered.
Instrumentation tape recorders are used in a variety of applications where the requirements is to record the information from transducers for subsequent analysis. Until now FM recording has reamined largely unchalleng...
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Instrumentation tape recorders are used in a variety of applications where the requirements is to record the information from transducers for subsequent analysis. Until now FM recording has reamined largely unchallenged in the field of instrumentation, partly because it is insensitive to varying tape parameters and can therefore provide more repeatable results. The price to pay for this increased performance is a narrow bandwidth: FM recorders cannot attain the same bandwidths as direct recording and so they consume more tape. Digital recording, on the other hand, is completely insensitive both to tape noise and transport speed variations: dynamic range is limited solely by electronic noise. Off-tape signals are identical to E-to-E signals, and figures of 96db are easily achieved. This represents at least two orders of magnitude increase in range over FM recording.
Quantization error is one of important measures that determine the performance of differential PCM. This paper investigates the DPCM quantization error characteristic and the exact determination of the stationary erro...
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Quantization error is one of important measures that determine the performance of differential PCM. This paper investigates the DPCM quantization error characteristic and the exact determination of the stationary error density. The input signal considered is a time-discrete increment process which is similar to the Wiener sequence, but with Laplacian increments. A system of equations satisfied by the characteristic function of the prediction error is derived and exact formulas are derived for error probability distributions. Using these distributions the mean-squared error, the entropy of the quantized outputs, and the probability of granular error are determined. The present analysis will be useful to performance analysis of DPCM with stationary non-Gaussian autoregressive inputs.
We analyze an improved soft-decision demodulation technique applied to mu -law pulse code modulation (PCM). The PCM-encoded waveforms are transmitted over Gaussian and Rayleigh fading channels by means of coherent pha...
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We analyze an improved soft-decision demodulation technique applied to mu -law pulse code modulation (PCM). The PCM-encoded waveforms are transmitted over Gaussian and Rayleigh fading channels by means of coherent phase shift keying modulation. Received bits which are unreliable are identified by means of soft-decision demodulation and erased. Each bit in the PCM word is assigned its own erasure threshold. The thresholds are optimized and are thus theoretically determined as a function of input power level to the quantizer, channel type and snr, and the relative mean-square error power that occurs when a PCM word containing an erased bit is estimated by means of prediction or interpolation.
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