Most of the technologically important aspects of magnetic recording are reviewed in a nonmathematical manner. After a discussion of recording media and heads, the fundamentals of the media magnetization and writing processes are considered. The factors limiting the bandwidth, distortion, and signal-to-noise ratio of audio, instrumentation, video, and digital recording machines are analyzed. The paper concludes with some predictions on the future of magnetic recording.

Tutorial review of magnetic recording

New applications such as Internet broadcast and communications, consumer multimedia products, digital AM broadcast and satellite networks are emerging. Those applications require moderate audio quality without annoying artifacts at bit rates below 16 kbit/s. Although speech coders provide high speech quality at bit rates around 8 kbit/s, they perform poorly when encoding audio signals. In this thesis, we present a novel transform coding paradigm based on the characteristics of the human hearing system. The proposed encoder, i.e., Narrowband Perceptual Audio Coder (NPAC), can accommodate a wide range of narrowband audio inputs without annoying artifacts at bit rates down to 8 kbit/s. NPAC employs a variety of algorithms to remove the perceptually irrelevant parts and statistical redundancies of the input signal. The new algorithms used in NPAC include a perceptual error measure in training the codebooks and selecting the best codewords, perceptually-based bit allocation algorithms and an adaptive predictive scheme to vector quantize the scale factors. The proposed encoder has moderate complexity and delivers good quality for narrowband audio inputs at around 1 bit/sample. Informal subjective tests have been conducted to compare the performance of NPAC with an 8 kbit/s commercially-available audio coder. The tests results show that NPAC performs better for both music and speech inputs.

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Perceptual Coding of Narrowband Audio Signals

A "hybrid" encoder has a characteristic action suitable for substantially full complementary playback in an audio signal transmission or recording system using decoders designed for one type complementary encode/decode system while permitting compatible decoding with decoders designed for use with another type complementary encode/decode system, or playback without any special decoders at all. The hybrid encoder is a series or parallel arrangement of two encoders that are modifications of standard encoders used in the two systems. In the preferred embodiments the two systems are A-type noise reduction and spectral recording. A variable degree of hybrid encoding is provided by varying the characteristic controlling circuit parameters in the two encoders, or in a preferred embodiment, by providing a variable combining arrangement that selects selects from the outputs of a hybrid encoder and an unmodified encoder (spectral recording).

Audio encoder for use with more than one decoder each having different characteristics

The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the apparatus comprises an upmix stage. The upmix stage is configured to generate the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

Audio signal of an fm stereo radio receiver by using parametric stereo

This paper investigates a hybrid scalar-vector quantization scheme for coding high quality audio signals. A wavelet packet transform (WPT) is used to decompose the audio signal into frequency bands slightly finer than the critical band divisions. A masking model computation is then used as input to the hybrid quantization scheme, where scalar quantization is used for coding the subbands from 0-5.5 kHz, and vector quantization is used for coding the subbands from 5.5-22 kHz. The performance of the proposed coder is assessed from segmental signal-to-noise ratios (SNR) and the perceived quality for a number of signals. The perceived quality is determined from informal comparisons between the uncoded signals at the original bitrate of 705 kb/s, and the same signals coded with (1) the proposed coder at 80 kb/s, (2) a coder using only scalar quantization at both 128 kb/s and 96 kb/s, and (3) the MPEG layer III coder at 64 kb/s. The comparisons indicate that a very good coder quality is possible with the proposed coder at bitrates of approximately 80 kb/s. This represents a saving of about 16 kb/s over full scalar quantization with a similar quality. Further bitrate reduction with the proposed coder is possible by entropy coding of the scalar quantized transform coefficients and the VQ indices.

David P. Robinson

Papers

AC-2 and AC-3: The Technology and Its Application

Production of Dolby B-Type Cassettes

Dolby B-Type Noise Reduction for FM Broadcasts

CP100 Cinema Processor--A New Audio Control Center for the Motion Picture Theatre