Pulse or digital communications – Bandwidth reduction or expansion
Reexamination Certificate
1999-02-09
2004-01-20
Chin, Stephen (Department: 2634)
Pulse or digital communications
Bandwidth reduction or expansion
C704S203000
Reexamination Certificate
active
06680972
ABSTRACT:
This application is the national phase under 35 U.S.C. §371 of prior PCT International Application No. PCT/IB98/00893 which has an International filing date of Jun. 9, 1998 which designated the United States of America.
TECHNICAL FIELD
In source coding systems, digital data is compressed before transmission or storage to reduce the required bitrate or storing capacity. The present invention relates to a new method and apparatus for the improvement of source coding systems by means of Spectral Band Replication (SBR). Substantial bitrate reduction is achieved while maintaining the same perceptual quality or conversely, an improvement in perceptual quality is achieved at a given bitrate. This is accomplished by means of spectral bandwidth reduction at the encoder side and subsequent spectral band replication at the decoder, whereby the invention exploits new concepts of signal redundancy in the spectral domain.
BACKGROUND OF THE INVENTION
Audio source coding techniques can be divided into two classes: natural audio coding and speech coding. Natural audio coding is commonly used for music or arbitrary signals at medium bitrates, and generally offers wide audio bandwidth. Speech coders are basically limited to speech reproduction but can on the other hand be used at very low bitrates, albeit with low audio bandwidth. Wideband speech offers a major subjective quality improvement over narrow band speech. Increasing the bandwidth not only improves intelligibility and naturalness of speech, but also facilitates speaker recognition. Wideband speech coding is thus an important issue in next generation telephone systems. Further, due to the tremendous growth of the multimedia field, transmission of music and other non-speech signals at high quality over telephone systems is a desirable feature.
A high-fidelity linear PCM signal is very inefficient in terms of bitrate versus the perceptual entropy. The CD standard dictates 44.1 kHz sampling frequency, 16 bits per sample resolution and stereo. This equals a bitrate of 1411 kbit/s. To drastically reduce the bitrate, source coding can be performed using split-band perceptual audio codecs. These natural audio codecs exploit perceptual irrelevancy and statistical redundancy in the signal. Using the best codec technology, approximately 90% data reduction can be achieved for a standard CD-format signal with practically no perceptible degradation. Very high sound quality in stereo is thus possible at around 96 kbit/s, i.e. a compression factor of approximately 15:1. Some perceptual codecs offer even higher compression ratios. To achieve this, it is common to reduce the sample-rate and thus the audio bandwidth. It is also common to decrease the number of quantization levels, allowing occasionally audible quantization distortion, and to employ degradation of the stereo field, through intensity coding. Excessive use of such methods results in annoying perceptual degradation Current codec technology is near saturation and further progress in coding gain is not expected. In order to improve the coding performance further, a new approach is necessary.
The human voice and most musical instruments generate quasistationary signals that emerge from oscillating systems. According to Fourier theory, any periodic signal may be expressed as a sum of sinusoids with the frequencies f, 2f, 3f, 4f, 5f etc. where f is the fundamental frequency. The frequencies form a harmonic series. A bandwidth limitation of such a signal is equivalent to a truncation of the harmonic series. Such a truncation alters the perceived timbre, tone colour, of a musical instrument or voice, and yields an audio signal that will sound “muffled” or “dull”, and intelligibility may be reduced. The high frequencies are thus important for the subjective impression of sound quality.
Prior art methods are mainly intended for improvement of speech codec performance and in particular intended for High Frequency Regeneration (HFR), an issue in speech coding. Such methods employ broadband linear frequency shifts, non-linearities or aliasing [U.S. Pat. No. 5,127,054] generating intermodulation products or other non-harmonic frequency components which cause severe dissonance when applied to music signals. Such dissonance is referred to in the speech coding literature as “harsh” and “rough” sounding. Other synthetic speech HFR methods generate sinusoidal harmonics that are based on fundamental pitch estimation and are thus limited to tonal stationary sounds [U.S. Pat. No. 4,771,465]. Such prior art methods, although useful for low-quality speech applications, do not work for high quality speech or music signals. A few methods attempt to improve the performance of high quality audio source codecs. One uses synthetic noise signals generated at the decoder to substitute noise-like signals in speech or music previously discarded by the encoder [“Improving Audio Codecs by Noise Substitution” D. Schultz, JAES, Vol. 44, No. 7/8, 1996]. This is performed within an otherwise normally transmitted highband at an intermittent basis when noise signals are present. Another method recreates some missing highband harmonics that were lost in the coding process [“Audio Spectral Coder” A. J. S. Ferreira, AES Preprint 4201, 100
th
Convention, May 11-14, 1996, Copenhagen] and is again dependent on tonal signals and pitch detection. Both methods operate at a low duty-cycle basis offering comparatively limited coding or performance gain.
SUMMARY OF THE INVENTION
The present invention provides a new method and an apparatus for substantial improvements of digital source coding systems and more specifically for the improvements of audio codecs. The objective includes bitrate reduction or improved perceptual quality or a combination thereof. The invention is based on new methods exploiting harmonic redundancy, offering the possibility to discard passbands of a signal prior to transmisson or storage. No perceptual degradation is perceived if the decoder performs high quality spectral replication according to the invention. The discarded bits represent the coding gain at a fixed perceptual quality. Alternatively, more bits can be allocated for encoding of the lowband information at a fixed bitrate, thereby achieving a higher perceptual quality.
The present invention postulates that a truncated harmonic series can be extended based on the direct relation between lowband and highband spectral components. This extended series resembles the original in a perceptual sense if certain rules are followed: First, the extrapolated spectral components must be harmonically related to the truncated harmonic series, in order to avoid dissonance-related artifacts. The present invention uses transposition as a means for the spectral replication process, which ensures that this criterion is met. It is however not necessary that the lowband spectral components form a harmonic series for successful operation, since new replicated components, harmonically related to those of the lowband, will not alter the noise-like or transient nature of the signal. A transposition is defined as a transfer of partials from one position to another on the musical scale while maintaining the frequency ratios of the partials. Second, the spectral envelope, i.e. the coarse spectral distribution, of the replicated highband, must reasonably well resemble that of the original signal. The present invention offers two modes of operation, SBR-1 and SBR-2, that differ in the way the spectral envelope is adjusted.
SBR-1, intended for the improvement of intermediate quality codec applications, is a single-ended process which relies exclusively on the information contained in a received lowband or lowpass signal at the decoder. The spectral envelope of this signal is determined and extrapolated, for instance using polynomials together with a set of rules or a codebook. This information is used to continuously adjust and equalise the replicated highband. The present SBR-1 method offers the advantage of post-processing, i.e. no modifications
Ekstrand Per Rune Albin
Henn Lars Fredrik
Kjörling Hans Magnus Kristofer
Liljeryd Lars Gustaf
Chin Stephen
Coding Technologies Sweden AB
Lugo David B.
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