Masahiro Iwadare

A bitrate and bandwidth scalable CELP coder

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.

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

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. >

Adaptive transform coding with an adaptive block size (ATC-ABS)

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.<<ETX>>

Separation of Small Particles Suspended in Liquid by Nonuniform Traveling Field

A study of size and charge-dependent separation of small particles in liquid using a traveling-field-type electric curtain device is made. The principle of the separation is to make use of the spatial harmonic components of the rotating traveling field produced by such a device, the first harmonic propagating in one direction, which plays a dominant role in the region distant from the electrodes, and the second harmonic propagating in the opposite direction, which becomes dominant near the electrodes. Small particles brought into this field undergo circular motion and, as a result of field nonuniformity, are repelled from the electrodes and drift in the direction of the dominant harmonics. The lighter or more charged particles are strongly repelled from the electrodes and swept by the first harmonic, while the heavier or less charged particles can approach the electrodes and are transported by the second harmonic in the opposite direction, thus enabling separation by mass and charge. First a theoretical investigation of this method is made to clarify the operation conditions for the separation, then the experimental observations of particle motion are made and scaling laws of transport velocity with the applied voltage and frequency are confirmed. Finally, an example of a cell separator design using this method is presented.

A single-chip MPEG/audio decoder LSI based on a compact decoding algorithm

A single-chip decoder LSI is developed for ISO/IEC MPEG (the International Organisation for Standardisation/the International Electrotechnical Commission, Moving Pictures Expert Group) audio. The applicable layers are Layer I and II of MPEG-1 and MPEG-2/Lower-Sampling-Frequency Mode. A fast calculation algorithm, which is also effective for on-chip memory reduction, is incorporated in audio signal synthesis. The reliability in bitstream synchronization is improved by including bitstream inconsistency detection. Bitstream error concealment by repeating previous audio data is supported. The decoding delay is adjustable when an optional external memory is connected to the LSI.

A new implementation of the Silicon Audio Player based on an MPEG/audio decoder LSI

A new implementation of the Silicon Audio player is presented. It decodes data which has been encoded by the MPEG/Audio Layer II algorithm standardized by the ISO (International Standardization Organization). The encoded data is stored in a semiconductor memory card. Decoding is carried out by an MPEG/audio decoder chip. Thanks to this dedicated LSI chip, the power consumption of the player is reduced so that the player could be driven by four nickel-metal hydride batteries for four and a half hours. Since the Silicon Audio player has no mechanical movement, it is robust against vibration that has been a serious problem for portable audio players. The recording time is defined as a function of the memory-card capacity and the compression ratio. Assuming one eighth compression with a sampling rate of 48 kHz, a 24-minute recording is possible with a 32-Mbyte memory card.

The Silicon Audio an audio-data compression and storage system with a semiconductor memory card

A new audio-data compression and storage system, the Silicon Audio, is presented. It employs the MPEG/Audio Layer II algorithm for data compression, which has been standardized by ISO (International Standardization Organization). A semiconductor memory card is equipped with to store the compressed signal. Decoding is carried out by a general purpose digital signal processor and a specially designed gate array chip. The package includes some special designs with vivid colors for the outdoor and sporting use. Since it has no mechanical movement, it is robust against vibration that has been a serious problem for portable audio players. The recording time is defined as a function of the memory-card capacity and the compression ratio. >

A robust 384 kbit/s stereo HiFi audio codec for ISDN applications

A robust stereo HiFi audio codec designed for 384 kb/s ISDN (integrated services digital network) subrate (H0) channel applications has been developed. The codec is based on subband ADPCM (adaptive differential pulse code modulation). In order to realize high coding quality, a new adaptive bit assignment function is introduced which needs no side-information transmission. Moreover, the coding algorithm brings about the robustness to withstand transmission bit errors in actual transmission environments such as satellite communications, where conventional adaptive bit assignment is inherently inferior. The codec is implemented using floating-point LSI signal processors. Subjective test results show that the coding quality is comparable to that of compact discs. The proposed algorithm can be applied to a basic ISDN channel by limiting the signal bandwidth to 15 kHz and transmitting FM-quality monophonic signals at 128 kb/s.<<ETX>>

An efficient tonal component coding algorithm for MPEG-2 Audio NBC

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.

The origin of digital information devices: the Silicon Audio and its family

This is an Open Access article, distributed under the terms of the Creative Commons Attribution-NonCommercial-NoDerivatives licence (http://creativecommons.org/licenses/by-ncnd/ 4.0/), which permits non-commercial re-use, distribution, and reproduction in any medium, provided the original work is unaltered and is properly cited. The written permission of Cambridge University Press must be obtained for commercial re-use or in order to create a derivative work. doi:10.1017/ATSIP.2017.16