Showing papers on "Adaptive Multi-Rate audio codec published in 1991"

16 kbps wideband speech coding technique based on algebraic CELP

Abstract: The application of algebraic code excited linear prediction (ACELP) coding to wideband speech is presented An algebraic codebook with a 20 bit address can be used without any storage requirements and, more importantly, with a very efficient search procedure which allows for real-time implementation The authors describe an efficient procedure for searching such a large codebook deploying a focused search strategy, where less than 01% of the codebook is searched with performance very close to that of a full search High-quality speech at a bit rate of 13 kbps was obtained >

Subband coding of stereophonic digital audio signals

Abstract: The exploitation of left-right correlation in a subband code for stereophonic audio signals is investigated. A transform of left and right signals into decorrelated intensity and error signals is presented. Although this can be seen as the optimal exploitation of redundancy, it yields only marginal gain in bit rate. If the reduced phase-sensitivity of the human observer can be exploited by encoding only the intensity signal, a substantial gain can be obtained. Preliminary results of a stereo codec are promising: at 192 kb/s good coding results have been obtained. >

Digital coding of high quality audio

Abstract: Based on frequency-domain techniques, coding of high-quality audio with bit rates down to 64 kbit/s is possible. This performance is achieved using perceptual coding. Transform coding can be used to get the best performance at very low bit rates. Real-time implementations of several types of low bit rate codecs have been developed. Standardization of low bit rate audio coding systems is under way in ISO/IEC JTC1/SC2/WG11 (MPEG/Audio). >

A High Quality Digital HDTV Codec

Abstract: A digital video codec has been developed for the Zenith/AT&T HDTV (high-definition TV) system for terrestrial broadcast over NTSC taboo channels. 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 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 combination of a sophisticated encoded video format and advanced bit error protection techniques results in a highly robust reception and decoding of the compressed video signal. >

High quality audio coding using analysis-by-synthesis technique

Abstract: A modified regular-pulse excitation (RPE) linear predictive coding (LPC) codec is used to encode audio signals of 15 kHz bandwidth. With appropriate perceptual masking and carefully selected excitation structure, a fixed-pulse excitation (FPE) LPC codec is able to provide high-fidelity audio output. The quantization of both the LPC parameters and the excitation pulse sequence is examined. It is found that the LPC parameters can be quantized efficiently by adaptive differential PCM, while the excitation pulse sequences need more bits to quantize the amplitudes compared to narrowband speech encoding. Informal listening tests show that high-fidelity audio can be achieved at bit rates less than 100 kb/s. >

A speech codec for cellular radio at a gross bit rate of 11.4 kb/s

Abstract: A speech codec, called TRPE-HLTP (transformed binary regular pulse excitation, high-resolution long-term prediction) has been developed. It has been submitted as a candidate for the half-rate codec in the GSM (Group Special Mobile) system and its gross bit rate is 11.4 kb/s. The speech coding algorithm is of the type often called analysis-by-synthesis linear prediction. Its key elements are LSF (line spectral frequency)-coded spectral parameters, a high resolution closed-loop adaptive codebook, and a speech-trained low-complexity transformed binary regular pulse innovation generator, resulting in a net bit rate of 6.9 kb/s. The channel coding scheme consists of forward error correction with convolutional encoding, interleaving, and error detection. >

A Multipulse Speech Codec for Digital Cellular Mobile Use

Abstract: There is much current interest in the development of a speech coding algorithm suitable for use in a digital cellular mobile radio telephone communications system. There are a number of potential advantages in converting the analogue FM cellular system in North America to a digital system: a) The capacity of the cellular system depends on spectrum utilization and the frequency reuse pattern. Conversion to digital transmission will initially increase the spectrum utilization by a factor of three. Digital speech compression allows three virtual channels to be transmitted over a single 30 KHz bandwidth radio channel using a TDMA (time-division multiple access) format. Improved voice coding techniques are expected to increase the capacity by an additional factor of two in the next few years. b) The digital transmission technique will support many new customer services such as data and FAX transmission, encryption and anti-fraud features. Additional services may be based on the ability to determine vehicle locations by triangularization from the base stations.

A motion vector replenishment video codec for ATM networks

Abstract: A low bit-rate video codec for ATM networks is described. It is based on two-layer coding principles. The base layer comprises the motion vectors plus a strip of interframe coded video data. The remaining video data are coded by a second layer. Transmission of the base layer cells is assumed to be guaranteed. The required guaranteed channel rate can be as low as that needed for a speech signal, such that networks like Orwell can handle them equally. The second layer cells may be lost, if congestion arises. Simulation results demonstrate the performance of the codec for a range of cell loss rates from the second layer.

Single board video codec for ISDN visual telephone

Abstract: A single board video codec for ISDN B/2B channel transmission, which depends on a CCITT standardization p*64 video coding algorithm and communication protocol, has been developed. The video codec is constructed with newly designed DSPs, four kinds of NTSC-CIF (Common Intermediate Format) mutual conversion LSIs, a transmission codec LSI and an AD/DA hybrid IC. The video codec codes and decodes a full CIF signal at a frame rate of 10 frames/s and communicates over either an ISDN 64 kb/s or 2*6 64 kb/s channel. The codec is fabricated on a single small board with a size of 280 mm*280 mm. Furthermore, a desk-top prototype visual telephone terminal which uses the video codec, voice codec and NCU has been developed. >

Hybrid coding with pre-buffering and pre-analysis in a software-based codec environment

Abstract: A concept for a software oriented video telephone codec compatible to CCITT draft recommendation H.261 is presented. An overview is given of the hybrid DPCM/transform coding which has been adopted for video telephone applications in digital networks at 64 kbit/s up to 2.048 Mbit/s data rate. The principle of an effective coding procedure is described leading to minimized and optimally distributed processing power while allowing high picture quality and minimum delay in order to retain interactivity of the video telephone service. The paper is focused on the coding strategy which is based on a pre-analysis of the input pictures in terms of change detection and structure analysis. Thus the processing power available can be efficiently utilized to get a constant picture quality at low hardware expense.

A high quality digital HDTV codec

Abstract: A digital video codec has been developed for the Zenith/AT&T HDTV system for terrestrial broadcast over NTSC taboo channels. 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 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 combination of a sophisticated encoded video format and advanced bit error protection techniques results in a highly robust reception and decoding of the compression video signal. >

Speech Coding Using Least-Squares Estimation

Abstract: An important goal in current speech coding research is providing high-quality speech at low bit rates (4.8–16 Kbps). Several methods [1]–[3] have been proposed recently to achieve this end. Compared to the conventional linear predictive (LP) vocoder [4], these methods employ an enhanced speech production model to synthesize speech. For example, instead of a single stage, the modulation filter now typically consists of two stages: i) a short-delay filter modeling the spectral envelope of speech, and ii) a long-delay filter modeling the spectral fine structure. Both are time-varying, all-pole filters and are derived from the original speech through LP analysis. Also, some information is provided about the excitation signal, which is selected by means of an analysis-by-synthesis procedure whereby a perceptually weighted error criterion is minimized In the multi-pulse linear predictive coder (MPLPC) [1], the excitation signal is a sequence of appropriately located and scaled impulses. In the code excited linear predictive coder (CELPC) [2], it is an entry from a codebook of white, gaussian noise sequences. In the self excited vocoder (SEV) [3], it is selected from the past history of the source excitation. As a result of these improvements, the above coders are able to synthesize high-quality speech at low bit rates.

A 6.3 kb/s CELP codec suitable for half-rate system

Abstract: A codec scheme suitable for the GSM (Group Special Mobile) half-rate system is proposed. The source coding has been performed with a CELP (code-excited linear prediction) algorithm, which provides good speech quality at low bit rates. In order to fit the recommended speech of 11.4 kb/s for the traffic channel, a 6.3 kb/s CELP codec and a FIRE protecting scheme adding 5.1 kb/s of redundancy were used. The real-time implementation is based on two PC boards connected via the PC bus: this makes it easy to test the system functions. The speech quality is almost the same as that of a full-rate codec on a wide range of experiments with different speakers and different languages. Channel error effects are considered. >

Bit-rate control algorithm on video packet coding and its characteristics

Abstract: A bit-rate control method based on a hybrid coding algorithm is proposed for packet video coding. Three types of bit-rate control modes, the constant bit-rate mode, the average bit-rate control mode and the free bit-rate mode, are investigated. According to the opinion test the latter two modes have an advantage in video quality over the constant bit-rate mode. A teleconference terminal, which includes this video codec, audio codec and packet adaptor, is implemented in actual hardware on High-speed Packet Switching system. On this system, video information are transmitted in UI frame (a packet format in X.25 protocol) at 64 K ∼ 800 Kbit/s.

The digital voice terminal for slow-FH radio application

Abstract: The authors present a real-time digital voice terminal (DVT) for slow frequency-hopping (slow-FH) radio applications to ensure a reliable radio link. Slow-FH technologies have been introduced for ECCM communications as a near-term solution which will not disturb the usage of the present conventional radio spectrum. Digital secure algorithms with digital speech codec are known to achieve a high degree of security. The DVT is composed of a speech codec and a voice-band modem. The speech codec that is used for the DVT is a 4.8 kbit/s pitch predictive coder using adaptive transform coding (PP-ATC). The modem is a QAM voice-band modem. This modem has a specific signal format to allow waveform shaping by a hopping transmitter to keep RF compatibility with conventional radio. The authors address aspects of the DVT architecture, the implementation of algorithms in the signal processors, and the system performance. >

New directions in low bit rate speech coding

Abstract: It is pointed out that impressive progress has been made during recent years in coding speech with high quality at low bit rates and at low cost. The new digital cellular system in North America is using coded speech at 8 kb/s, but much lower bit rates are needed to support the increasing demand for cellular telephones. It is noted that incremental changes in the present technology are unlikely to produce high-quality speech at very low bit rates. Current research at Bell Laboratories is aimed at creating new representations of speech, both in frequency and time, to allow for efficient coding of speech information. The author reviews the key ideas that support the present speech coding technology and discusses some of the promising new directions. >

Evaluation system for mobile radio speech codec

Abstract: A multifunctional speech quality evaluation system has been developed to test the performance of mobile radio speech codecs. This system is characterized by a simple hardware configuration, a variety of software functions, and general interface specifications. Major degradation factors in mobile radio communications are taken into account, e.g., Rayleigh fading channel errors and background noise. The effects of PSTN, including transmission loss, noise, delay, and echo, are also considered. This evaluation system was used in the full-rate speech codec standardization for the Japanese digital cellular system in 1990. >

Real-time implementation of a high quality CELP codec for digital mobile radio system

Abstract: The authors describe a real-time implementation of a voiceband codec suitable for the GSM Half-Size Digital Mobile Radio (DMR) system at 900 MHz. This codec exploits the CELP (code excited linear prediction) algorithm running at 6.3 kbit/s in connection with a channel coding scheme, adding 5.1 kbit/s of redundancy to reach the final rate of 11.4 kbit/s for the traffic channel. A comparative evaluation between the full-rate and the half-rate source codecs is presented: the speech quality is almost the same on a wide range of experiments and different speakers. >