Kohei Iseda

Speech encoding/decoding apparatus having selected encoders

Several encoders perform a local decoding of a speech signal and extract excitation information and vocal tract information from a speech signal for an encoding operation. The transmission rate ratio between the excitation information and the vocal tract information are different for each encoder. An evaluation/selection unit evaluates the quality of decoded signals subjected to a local decoding in each of the encoders, determines the most suitable encoders from among the several encoders based on the result of the evaluation, and selects the most suitable encoder, thereby outputting the selection result as selection information. The decoder decodes a speech signal based on selection information, vocal tract information and excitation information. The evaluation/selection unit selects the output from the encoder in which the quality of a locally decoded signal is the most preferable. When vocal tract information changes little, the vocal tract information is not output, thereby allowing for increased quality of information. As much of the surplus of unused vocal tract information as possible is assigned to a residual signal. Thus, the quality of a decoded speech signal is improved.

ADPCM with a multiquantizer for speech coding

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding. >

A high-efficiency speech coding algorithm based on ADPCM with multi-quantizer

Adaptive differential PCM (ADPCM) is an effective coding scheme to simplify the hardware and shorten the processing delay to realize a high-efficiency speech codec. The ADPCM with Multi-Quantizer (ADPCM-MQ) coding has been proposed as one of the highly efficient coding methods. In the ADPCM-MQ codec several ADPCM coding blocks with different quantization step-size update rates are operated in parallel, and the quantizer that gives the best characteristics is found and selected dynamically for each frame. This paper describes a new 8 to 9.6 kbps ADPCM-MQ coding algorithm that includes tree coding to improve the per-sample quantizing characteristic, and sub-band coding with high frequency band reconstruction to realize a lower bit rate coding. Computer simulation indicates good quality for speech reproduced by this algorithm with a segmental signal-to-noise ratio of 14 to 15 dB. Adaptive postfiltering can be added to enhance the subjective characteristics of the reproduced speech.

CORBA-based network operation system architecture

The scope of network management systems has been expanding rapidly, from simple network element management systems to integrated management systems that include network management and service management. Since the telecommunications world has become an open market, sales competition has intensified. Accordingly, a flexible, extendable, and reusable network management system is required. A distributed object-oriented environment meets these requirements, and CORBA (Common Object Request Broker Architecture) is a promising candidate as a distributed environment. An integrated management system should provide management capabilities that are independent from management protocols and from the location of Managed Objects (MOs) which represent managed resources. The system should also be able to execute multiple management operations (multiple management scenarios) concurrently. In this paper, we describe a technique of introducing an interface for each MO, called a Proxy MO, that encapsulates the management protocols and locations of the MOs and which are concurrently controlled in the CORBA environment. This technique provides a flexible, extendable, and reusable network management system.

A 16 kbps ADPCM with multi-quantizer (ADPCM-MQ) codec and its implementation by digital signal processor

This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance. The coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ) which selects the optimum ADPCM coder frame by frame and switches to it dynamically. To implement this coding algorithm, we used two Fujitsu DSPs (MB8764), 1.5 chips for the encoder and 0.5 chip for the decoder. 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 to 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 ms, no echo-canceller is needed. Moreover, 2400 bps voice band data (CCITT Rec.V. 26) could be transmitted with a data error rate from 10^-7 to 5×10^-6, and G.III facsimiles were successfully transmitted using this codec.

An implementation of a variable rate codec based on ADPCM with multi-quantizer (ADPCM-MQ)

Discusses an implementation of an experimental variable bit rate codec (VRC) based on adaptive differential pulse code modulation with a multiquantizer (ADPCM-MQ) that processes speech signals at 16 kb/s to 48 kb/s. The authors propose a method for the combined use of constant SNR (signal/noise ratio) and constant noise to determine optimum bit/rate control of the VRC. This algorithm was implemented using six digital signal processor chips. The average bit rate of the prototype codec for the telephone voice channel was about 23 kb/s, including supplementary information, and a quality better than that specified in G.721 (32 kb/s ADPCM) was observed.<<ETX>>

Modeling of self-healing managed object for TMN

In high capacity optical fiber networks, fast restoration from catastrophic failure is increasingly more important. A fast restoration self-healing algorithm and an optimized network design method are illustrated. The proposed telecommunications managment network (TMN) is analyzed from the standpoint of implementing the self-healing algorithm. A design and an effective implementation scheme of the self-healing in accordance with TMN are proposed.<<ETX>>

An 8 kbps TC-MQ (time domain compression ADPCM-MQ) speech codec

A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed. It is based on time scale modification and sub-band coding with the aid of ADPCM with a multiquantizer. For time scale modification, the decimation/interpolation technique is introduced for the unvoiced period. Furthermore, in order to get computational accuracy of the pitch extraction for the voiced period, a method using the normalized autocovariance function is proposed. The algorithm was confirmed by computer simulations. The segmental SNR was about 13-16 dB for Japanese short sentences. A good mean opinion score value was also obtained by means of a subjective evaluation test.<<ETX>>