Showing papers on "Adaptive Multi-Rate audio codec published in 1986"

Coding speech at low bit rates: Advanced algorithms and hardware for voice telecommunications are paring hit rates by at least a factor of four, without losing intelligibility

Abstract: Digital coding techniques are described that promise to enhance the applicability of voice communications and storage. The techniques allow more speech to be represented with a given number of binary digits, without losing natural voice quality. The advanced coding techniques just becoming available yield natural-sounding telephone speech at digital transmission rates of 16, 8, and eventually 4 Kb/s as well as the standard 64 and 32 Kb/s rates. The author discusses quantification of voice quality, the implementation of algorithms for digital coding using chips, algorithms for prediction and bit allocation, human sound perception, and hybrid coding approaches.

A study on the realtionships between stochastic and harmonic coding

Abstract: Two of the coding methods which have appeared recently, attempting to solve the problem of high quality speech coding at bit rates below 9.6 kbits/sec are harmonic and stochastic coding. The two represent distinct approaches to this problem and yield synthetic speech of a very different kind. In this paper, we discuss the advantages and disadvantages of both methods for processing voiced and unvoiced speech segments, and examine the potential benefits of combining the two methods in coding applications.

A Motion-Compensated Interframe CODEC

Abstract: This paper describes an adaptive intra-interframe codec with motion-compensation followed by an entropy coding for prediction error signal as well as for motion vector information. This adaptive prediction is highly efficient even for very fast motion as well as scene change where motion compensation is ineffective. Prediction error and vector information are code-converted for transmission by means of an entropy coding where contiguous zero signal is run-length coded and non-zero signal is Huffman-coded. Based upon the algorithms described in this paper a practical codec has been developed for videoconference use at sub-primary rate. According to a brief subjective evaluation, the codec provides good picture quality even at a 384 kb/s transmission bit rate.

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

Abstract: 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.

An advanced 32 kbit/s ADPCM coding to transmit speech and high-speed voiceband data

Abstract: This paper describes a performance evaluation of an advanced 32 kbit/s ADPCM hardware codec in terms of speech and high-speed voiceband data, V.27 and V.29 at 4.8, 7.2 and 9.6 kbit/s. The results indicated that the advanced 32 kbit/s ADPCM provided an equivalent subjective speech quality to that of the G.721 ADPCM recommended by the CCITT, and was able to maintain the transparency of these high-speed voiceband data at 7.2 kbit/s over at least four and at 9.6 kbit/s over at least two asynchronous tandem links, respectively. Furthermore, the advanced ADPCM can quickly recover a complete synchronous operation in the tandem connections, as burst errors or short-hits damage the digital links.

CCITT Standardizing activities on speech coding

Abstract: CCITT (International Telegraph and Telephone Consultative Committee) Study Group XVIII is in charge of studies on speech processing, such as speech coding techniques. During its last study period (1981-84), a recommendation for 32 kbit/s speech coding was made. Afterwards, Study Group XVIII began standardization studies on wideband speech coding (i.e., 7 kHz audio coding) at 64 kbit/s or less. This paper reports on the progress of these standardization studies. The 32 kbit/s Adaptive Differential PCM (ADPCM) standard is briefly reviewed. Also, the 7 kHz audio coding at 64 kbit/s or less is mainly discussed focussing on its applications, design requirements and coding algorithm.

Implementation of a 32 Kbit/s ADPCM Codec Using a General-Purpose Digital Signal Processor

Abstract: This paper describes an implementation of a CCITT G.721 compatible 32kbit/s ADPCM codec, using a general-purpose digital signal processor FDSP-3 (MB8764). A single-channel ADPCM codec is realized by two FDSP-3 chips-one for the encoder and the other for the decoder. Meticulous programming techniques are employed to achieve exact computation of the CCITT algorithm exploiting all the available resources of the 16-bit fixed-point DSP. It is shown that the whole codec computation can be accomplished in about 2350 machine cycles. Thus, two FDSP-3 chips operating at 10 MHz machine cycle can handle the whole computation. The paper also covers the comparison of straight fixed-point format and the G.721 realization, and briefly examines the compatibility issue between these two methods.

Performance of 16 kbit/s APC codec with maximum likelihood quantization for low-speed voiceband data

Abstract: This paper presents an evaluation of the performance, with low-speed voiceband data at 300 bit/s, 1.2 kbit/s and 2.4 kbit/s, of a 16 kbit/s APC (Adaptive Predictive Coding) with maximum likelihood quantization(MLQ). By measuring modem bit error rates, the effect of a long-term predictor and influence of a subscriber line and channel bit errors over the APC-MLQ codec were investigated. The results indicate that the 16 kbit/s APC-MLQ can successfully transmit these low-speed voiceband data and is applicable to digital maritime satellite communication systems which must carry both speech and voiceband data up to 2.4 kbit/s. The performance has also been confirmed through field trials with G-III facsimile transmission at 2.4 kbit/s over a digital maritime satellite transmission system.

Single DSP 8kbps speech CODEC

Abstract: This paper describes the implementation of an 8kbps speech CODEC using a single DSP. First, an effective algorithm based on the TOR (Thinned-Out Residual) method has been developed by modifying the means to thin out and interpolate residuals in voiced and unvoiced frames, and to extract and modify the pitch period. Then, a compact CODEC which consists of one DSP and one MPU each has been developed using the new algorithm. This CODEC supports not only speech coding functions, but also frame synchronization functions and echo suppressing functions. The measured speech quality is quite satisfactory.

A 32kbps ADPCM with improvement in coding characteristics for 9600bps modem signal

Abstract: This paper describes a new 32 kbps ADPCM CODEC which shows high-quality speech coding characteristics and data transmission capability up to 9600 bps. The proposed ADPCM CODEC consists of two coding modes. The first coding mode can code the speech signal and the data transmission signal up to 4800 bps by using the coding algorithm of CCITT recommendation G.721. The second coding algorithm is tuned to handle the 9600 bps modem signal. These two coding modes are automatically switched depending on the input signal to the CODEC. The switching algorithm is based on detection of modem training signal and is found to be highly reliable in practical environments. This algorithm thus provides a means to be compatible with the standard CCITT algorithm for speech and data transmission up to 4800 bps and at the same time be transparent to data transmission at 9600 bps.

Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)

Abstract: Recently much intensive research of 16kbps Speech coding algorithm has been conducted aiming to reduce the transmission bit rate and yet provides high speech quality. Adaptive predictive coding with adaptive bit allocation (APC-AB)[1] is considered to be one promising approach. However, the processing of this coding algorithm is so complicated that the implementation of the algorithm on a general-purpose signal processor, especially if fixed-point arithmetic DSPs are used, requires careful study of arithmetic operation precision and same way to reduce the number of processing cycles. Taking account of these points, real-time signal processing using a fixed-point signal processing chip (FDSP-3) has been studied, and a prototype codec has been realized. The prototype codec satisfied the CCITT mask of signal-to-total distortion ratio for PCM codecs and showed quality good enough for "toll" speech.

Wideband speech codec at 64kb/s using an advanced digital signal processor "DSSP-1"

Abstract: A 7 kHz-band speech CODEC using an advanced digital signal processor is developed. The coding algorithm for this CODEC requires greater processing power due to its complexity and high sampling rate. A high-performance digital signal processor VLSI, "DSSP-1", is applied to this CODEC. Arithmetic word-length, arithmetic operation speed, and memory capacity required for this CODEC is evaluated. DSSP-1 can satisfy these requirements; this CODEC has been built with two DSSP-1 chips and several memory chips.