This paper proposes a flexible CELP speech coder with bitrate and bandwidth scalabilities for multimedia applications. The coder is based on multi-pulse-based CELP coding and consists of a bitrate scalable base-band coder and a bandwidth extension tool. The bitrate scalable base-band CELP coder employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. The bandwidth scalability is realized by bandwidth conversion from base-band CELP parameters to those for wideband without a widely used subband structure. The bandwidth conversion improves base-band coding quality and expands bandwidth, simultaneously. The comparison test results show that the bitrate scalable coder is equivalent in speech quality to the fixed-bitrate CELP coder at the same bitrate for the narrowband speech. In the mean opinion score (MOS) tests, the proposed 16 kbit/s coder with the bandwidth scalability achieves equivalent coding quality to ITU-T G.722 at 56 kbit/s. The proposed coder is currently evaluated as the MPEG-4 CELP speech standard.

This paper proposes a tonal component coding algorithm for a codec that employs a transform followed by Huffman coding, such as MPEG-2 Audio NBC (non-backward compatible). After the input audio signal is mapped onto a frequency domain, the proposed algorithm withdraws local maximum components that degrade coding efficiency. By this withdrawal, the flatness of the spectrum increases and the efficiency in Huffman coding is improved. The withdrawn components are encoded separately as side information. When the frequency resolution of the time/frequency mapping is high, this algorithm works more effectively since local maximum samples appear more frequently with such a mapping. Simulation results show that this algorithm achieves as much as 11% bit reduction per frame and improves the coding efficiency in 41% of all the audio frames.

Two kinds of compact and motor-less multimedia players are developed employing PC memory cards as the storage device. The VHS-quality video and the compact-disc quality sound play is supported. The MPEG-1 coding is incorporated for video and audio compression. Since the compressed information is stored in memory LSIs and moving mechanical components are not required, lightweight, shock proof and high-reliability are achieved. Instantaneous random program access is also available. A portable model, "Silicon View Player" equips with a 2.5-inch liquid crystal display panel and weights 295 g with a rechargeable battery. "Shopping Navigation" with a larger Pinch panel is a shopping guide terminal with an additional user-interactive function for sales promotion at retail stores. These models will pave the way for portable and business-use multimedia applications which do not use electro-magnetic storage devices.

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.>

A coding technique is presented for high-quality audio signals based on adaptive transform coding (ATC). Adaptive block size selection by the proposed algorithm ensures an appropriate block size resulting in improved SNR (signal-to-noise ratio) for a wide variety of source signals. A feedback approach, based on SNR, and a feedforward approach, based on interblock differences in input time-domain samples, to adaptive block size assignment are proposed and evaluated. Computer simulation results show that average segmental SNR by the feedback approach is improved by as much as 4.8 dB over the conventional fixed-block-size ATC. The feedforward approach is realized with much-simplified hardware; nevertheless, its SNR degradation from that by the feedback approach is 1.6 dB, even in the worst case. Both approaches are successful in pre-echo suppression to a satisfactory level. Time-domain aliasing cancellation has the potential to increase the superiority of the new algorithm.>

Two simple 64-kb/s wideband coding approaches using 32-kb/s ADPCM (adaptive digital pulse-code modulated) channel banks are proposed and compared to CCITT 64 kb/s ADPCM, which is being recommended as CCITT G.722. These two, folding ADPCM and QMF ADPCM, are intended to pave the way for smooth transition from conventional 4-kHz band telephone systems to 7-kHz wideband systems in private networks. The first approach, supporting the high-quality audio program transmission, requires only samplers and multiplexers at the input and output ports of the channel banks. In the second approach, samplers and multiplexers are replaced by quadrature mirror filters in order to increase coding quality. Performance test results for audio signal transmission show that these simplified approaches provide an inexpensive way to introduce wideband communication systems.>