Showing papers on "Adaptive Multi-Rate audio codec published in 1992"

A 128 Kb/s Hi-Fi audio codec based on adaptive transform coding with adaptive block size MDCT

Abstract: A Hi-Fi audio codec with an improved adaptive transform coding (ATC) algorithm is presented using digital signal processors (DSPs). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for preecho suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The codec was realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the codec show that the coding quality is comparable to that of compact disc signals. >

Bit rates in audio source coding

Abstract: The goal is to introduce and solve the audio coding optimization problem. Psychoacoustic results such as masking and excitation pattern models are combined with results from rate distortion theory to formulate the audio coding optimization problem. The solution of the audio optimization problem is a masked error spectrum, prescribing how quantization noise must be distributed over the audio spectrum to obtain a minimal bit rate and an inaudible coding errors. This result cannot only be used to estimate performance bounds, but can also be directly applied in audio coding systems. Subband coding applications to magnetic recording and transmission are discussed in some detail. Performance bounds for this type of subband coding system are derived. >

Coding of wideband speech

Abstract: The technologies of ISDN teleconferencing, CD-ROM multimedia services, and High Definition Television are creating new opportunities and challenges for the digital coding of wideband audio signals, wideband speech in particular. In the coding of wideband speech, an important point of reference is the CCITT standard for 7 kHz speech at a rate of 64 kbit/s. Results of recent research are pointing to better capabilities — higher signal bandwidth at 64 kbit/s, and 7 kHz bandwidth at lower bit-rates such as 32 and 16 kbit/s. The coding of audio with a signal bandwidth of 20 kHz is receiving significant attention due to recent activity in the ISO (International Standards Organization), with a goal of storing a CD-grade monophonic audio channel at a bit-rate not exceeding 128 kbit/s. Prospects for accomplishing this are very good. As a side result, emerging algorithms will offer very attractive options at lower rates such as 96 and 64 kbit/s. As we address new challenges in wideband speech technology, several strides in coding research are likely to occur. Among these are refinements of existing models for auditory noise-masking, and a unification of linear prediction and frequency-domain coding.

A codec for HDTV

Abstract: A high-quality digital video codec has been developed for the Zenith/AT&T HDTV system. It adaptively selects between two transmission modes with differing rates and robustness. The codec works on an image progressively scanned with 1575 scan lines every 1/30th of a second and achieves a compression ratio of approximately 50 to 1. The high compression ratio facilitates robust transmission of the compressed HDTV signal within an NTSC taboo channel. Transparent image quality is achieved using motion compensated transform coding coupled with a perceptual criterion to determine the quantization accuracy required for each transform coefficient. The codec has been designed to minimize complexity and memory in the receiver. >

Manchester encoder–decoder for optical data communication links

Abstract: A new encoder-decoder (CODEC) design of a Manchester coding scheme suitable for optical data communication links is presented. The design is simple and uses off-the-shelf digital electronic components and subsystems. The CODEC can be used for high data rate transmissions, typical of opticalfiber systems and local area networks. The decoder is insensitive to variations in the clock rates within the range of +/-33%, whereas the encoder, which is a simple XOR logic gate, is not affected by clock variations. During high-frequency operation (e.g., at 100 MHz), the CODEC can be operated at a wide range of frequencies (from 66.6 to 133.3 MHz) without modification to the CODEC circuitry. Furthermore, the CODEC can be made to operate at any data rate by a simple change of a single capacitor or a single resistor in the decoder circuit. The CODEC was built in the laboratory by using transistor-transistor logicintegrated circuits. It was experimentally found that with this decoder the transmitted data, as well as the cloc, can be recovered from the Manchester coded signal without being affected by clock variations within the designed range.

Standardization activities on 8-kbit/s speech coding in CCITT SGXV

Abstract: The CCITT (International Telegraph and Telephone Consultative Committee) Study Group XV is in charge of studies on speech processing. SGXV has already drafted standard G.728 for a 16-kbit/s low-delay speech coding algorithm and is now studying 8-kbit/s speech coding algorithms. The paper reviews the recent activities of SGXV/WP2 on 8-kbit/s speech coding algorithm. These activities include discussion of the algorithm's application, of performance requirements and objectives, and of the procedure for evaluating the candidate algorithms. It also introduces one of the probable algorithms and presents this working group's plans for future work. >

An overview of speech coding techniques

Abstract: This brief introduction to speech coding algorithms in general gives emphasis to the encoding of narrowband-telephone signals for network/communication quality speech. The most important coding techniques and achievements are discussed together with existing international coding standards. Future directions are considered in terms of foreseen speech technology applications and the coding algorithms needed to support these.

A combined CELP/Reed-Solomon codec for mobile radio applications

Abstract: The performance of an error-protected speech codec for mobile radio applications is investigated. The speech codec is a 4 kb/s variation of the proposed Federal Standard 1016 CELP. The major difference is the use of a trained codebook (as opposed to a stochastic codebook in the proposed standard). The channel codec on the other hand, consists of a bank of rate-compatible punctured Reed-Solomon (RS) codes. The two subsystems are combined in an optimal fashion, according to the sensitivity of the speech elements. This implies that the most sensitive bits are protected by the most powerful RS codes, while the least sensitive bits are (either uncoded or) protected by the least powerful code. The performance of such a combined codec under different sets of system parameters and channel conditions is studied. In all cases, though, the aggregate rate is fixed at about 6.4 kb/s. The simulation results show that under the condition that there is no interleaving delay, the resulting codec can provide good quality speech at a channel bit error rate as high as 10/sup -2/. >

Half rate CELP codec candidate for North American digital cellular systems

Abstract: A half rate speech/channel codec intended for the TIA technology assessment is described. The requirements for this assessment are an overall bit rate of 6.4 kbps and a frame size of 40 msec. The codec design and its performance is presented. >

Testing of video codecs with a three-dimensional test signal

Abstract: A three-dimensional test signal for a video codec(14) is generated by injecting a conventional foreground test signal at a predetermined location as a foreground component within a background test signal as a background component, the background component having a variable complexity. The three-dimensional test signal is input to the video codec and the output of the video codec is measured using conventional measurement instruments(16). The background component may be a pseudo-random noise signal, a zone plate signal or other variable complexity-type signal, with the foreground component occurring at greater intervals than the neighboring pixels used by the codec compression algorithm. The conventional measurement instruments display only the foreground component with distortions in the video codec caused by the complexity of the background component appearing in the display. A spectral display of the output of the video codec using a zone plate signal as the background component may also be used to characterize the video codec performance.

Design of a H.261 video codec with 12 Xilinx LCAs

Abstract: The authors describe the design of a video codec CCITT H.261 on 2 PC/AT boards based on special purpose DSPs, RAMs and Xilinx LCAs. The CCITT H.261 recommendations describes the video coding and decoding methods for the moving picture component of audiovisual services at the rates of p*64 kbit/s, where p is in the range 1 to 30. The codec has been developed jointly with Centre National d'Etudes des Telecommunications and is now connected to ISDN for videophone and videoconferencing communications. In addition to describing the architecture and design, the authors discuss the development system used in the implementation. >

ATC-based hi-fi audio coding for ATM networks

Abstract: The authors propose an adaptive transform coding (ATC)-based hi-fi audio coding method for asynchronous transfer mode (ATM) networks. To maintain robustness despite cell loss, a novel layered coding scheme has been employed in ATC. In this approach, higher/lower priority is assigned to most/least significant bits, rather than to lower/higher frequency bands, as is commonly done in video coding. The number of bits for most/least significant bits is determined on the basis of two different coding-rates. Subjective test results for 192 kbps/96 kbps coding show that, even under conditions of cell loss, it is possible to maintain a level of quality equivalent to 192 kbps coding quality. The proposed approach appears to be promising for high-quality hi-fi audio signal transmission in ATM networks. >

Half-rate voice coding system for mobile radio

Abstract: A novel speech compression method and its hardware implementation which provide significant improvement in spectrum efficiency for land mobile radio systems are described. The method involves the use of the adaptive selection scheme of the DCT coefficients in conjunction with the pitch prediction to achieve excellent speech quality and low complexity at rates of 5.6 kb/s. The computational simplicity of the coding algorithm makes it particularly well suited to real-time implementation of a significant lower cost than LPC based systems producing similar speech quality. It is seen that DSP realization of the coder results in a viable prototype implementation yielding good speech quality. >

HDTV: a 140 Mbit/s coded for 'contribution' quality coding

Abstract: Digitising the luminance and chrominance components of an HDTV signal results in a bit rate of about 1.2 Gbit/s. For long-distance transmission an obvious goal is to reduce this bit rate by a factor of about 10 in order to fit into the current standard bit rate of 140 Mbit/s. In order to maintain contribution quality for such bit-rate reduction factors, it is necessary to use a combination of techniques such as transform coding, motion-compensated interframe prediction and variable-length coding. The BBC, as part of the RACE HIVITS Project (No. 1018) has participated in the construction of a such a 140 Mbit/s codec for HDTV. This codec is based around a codec, that has also been developed within the HIVITS project and manufactured by Thomson, for the coding of conventional-definition television within 34 Mbit/s.

VLSI video codec using programmable DSP

Abstract: A video codec based on CCITT's standardized p*64 video coding algorithm and communication protocol has been developed for ISDN H0 rate transmission. The codec is fabricated on two 280-mm*280-mm boards and is composed of new DSPs, four kinds of NTSC-CIF (National Television Committee-Common Intermediate Format) mutual conversion LSIs, a transmission codec LSI, and an AD/DA (analog-to-digital/digital-to-analog) hybrid IC. The codec codes and decodes full CIF signs at a rate of 15 frames/s and communicates at an ISDN 384-kb/s rate. Application programs for the codec, such as picture reconstruction capability driven by DSPs in the coder or decoder, can be performed by using residual DSP processing power without the need for any hardware addition. >

Real-Time Implementation of a CELP Codec with Unequal Error Protection

Abstract: A digital speech codec for mobile satellite communication has been developed. It includes both source and channel coding on two digital signal processors. A CELP (code-excited linear predictive) speech coder has been combined with a channel coder using RCPC (rate compatible punctured convolutional) codes providing an unequal error protection. The paper describes the structure of the CELP codec, the assessment of the bit error sensitivity, the choice of suitable channel codes, and the real-time hardware concept.