Sound quality of established low bit-rate audio coding...

Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion

Reexamination Certificate

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Details

C704S501000, C704S502000

Reexamination Certificate

active

06226616

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to low bit-rate audio coding systems and more specifically to a method of improving sound quality of established low bit-rate audio coding systems without loss of decoder compatibility.
2. Description of the Related Art
Numbers of low bit-rate audio coding systems are currently in use in a wide range of consumer and professional audio playback products and services. For example, Dolby AC3 (Dolby digital) audio coding system is a world-wide standard for encoding stereo and 5.1 channel audio sound tracks for Laser Disc, NTSC coded DVD video, and ATV, using bit rates up to 640 kbit/s. MPEG I and MPEG II audio coding standards are widely used for stereo and multi-channel sound track encoding for PAL encoded DVD video, terrestrial digital radio broadcasting in Europe and Satellite broadcasting in the US, at bit rates up to 768 kbit/s. DTS (Digital Theater Systems) Coherent Acoustics audio coding system is frequently used for studio quality 5.1 channel audio sound tracks for Compact Disc, DVD video and Laser Disc and bit rates up to 1536 kbit/s.
A major problem with these systems is that their designs are inflexible in that they cannot be easily upgraded to accommodate higher PCM sampling frequencies, PCM word lengths or higher system bit rates. This will become an important issue in coming years as the music and film industry moves to drop the old compact disc digital audio format of 44.1 kHz sampling frequency and 16 bit word length, and adopt the new DVD audio PCM mastering format of 96 kHz sampling and 24 bits word length.
As a result, audio delivery using existing audio encoding systems such as AC-3, MPEG and DTS, must adapt to allow the benefits of this increased signal fidelity to pass to the consumer. Unfortunately a large installed base of audio decoder processing chips (DSPs) which implement these decoder functions already reside in the existing consumer base. These decoders cannot be easily upgraded to accommodate the increasing sampling rates, word size, or bit rates. Consequently, music and film content providers selling product through these mediums will be forced to continue to supply coded audio streams that are compliant with the old standards. This implies that in the future, delivery media such as DVD audio, ATV, satellite radio etc, may be forced to deliver multiple bit streams, each conforming to different standards. For example, one stream would be included in order to allow owners of existing playback systems to receive and play the standard audio tracks, while a second stream would also reside to allow owners of newer equipment to play audio tracks encoded using the 96 kHz/24 bit PCM format and take advantage of the inherently higher fidelity.
The problem with this method of delivery is that many of the playback mediums may not be able to afford the extra bandwidth, or channel capacity, necessary to send the additional audio streams. The bit rate of the additional bit streams (for example, those that support 96 kHz/24 bits) will be at least equal or greater than those that support the old format. Hence the bit rate will most likely double or more, in order to support two or more audio standards.
SUMMARY OF THE INVENTION
In view of the above problems, the present invention provides a coding methodology that extends the frequency range and lowers the noise floor while avoiding having to deliver replica audio data and is therefore much more efficient at accommodating changes in PCM sampling frequency, word length and coding bit rates.
This is accomplished with a ‘core’ plus ‘extension’ coding methodology, in which the traditional audio coding algorithm constitutes the ‘core’ audio coder, and remains unaltered. The audio data necessary to represent higher audio frequencies (in the case of higher sampling rates) or higher sample resolution (in the case of larger word lengths), or both, is transmitted as an ‘extension’ stream. This allows audio content providers to include a single audio bit stream that is compatible with different types of decoders resident in the consumer equipment base. The core stream will be decoded by the older decoders which will ignore the extension data, while newer decoders will make use of both core and extension data streams giving higher quality sound reproduction.
A key feature of the system is that the extension data is generated by subtracting a reconstructed core signal (encoded/decoded and/or downsampled/upsampled) from the original ‘high fidelity’ input signal. The resulting difference signal is encoded to produce the extension stream. With this technique, aliasing fold-back into either the core or extension signals is avoided. Hence, the quality of the core audio is unaffected by the inclusion of the extension stream. For the system to work in its most elementary mode, only the latency, or delay, of the core coder needs to be known. Hence, this method can be successfully applied to any audio coding system even without knowledge of the coder's internal algorithms or implementation details. However the system can be made to work more efficiently if the extension coder is design to match the core coder over the frequency range of the core signal.
These and other features and advantages of the invention will be apparent to those skilled in the art from the following detailed description of preferred embodiments, taken together with the accompanying drawings, in which:


REFERENCES:
patent: Re. 32124 (1986-04-01), Atal
patent: 4354057 (1982-10-01), Atal
patent: 4554670 (1985-11-01), Aiko et al.
patent: 4860312 (1989-08-01), Heuvel et al.

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