Electrical computers and digital data processing systems: input/ – Input/output data processing – Input/output data buffering
Reexamination Certificate
1997-05-02
2001-02-20
Harvey, Minsun Oh (Department: 2747)
Electrical computers and digital data processing systems: input/
Input/output data processing
Input/output data buffering
C710S056000, C700S094000
Reexamination Certificate
active
06192427
ABSTRACT:
FIELD OF THE INVENTION
This invention relates in general to the field of electronic systems and more particularly to an improved modular audio data processing architecture and method of operation.
BACKGROUND OF THE INVENTION
Audio and video data compression for digital transmission of information will soon be used in large scale transmission systems for television and radio broadcasts as well as for encoding and playback of audio and video from such media as digital compact cassette and minidisc.
The Motion Pictures Expert Group (MPEG) has promulgated the MPEG audio and video standards for compression and decompression algorithms to be used in the digital transmission and receipt of audio and video broadcasts in ISO-11172 (hereinafter the “MPEG Standard”). The MPEG Standard provides for the efficient compression of data according to an established psychoacoustic model to enable real time transmission, decompression and broadcast of CD-quality sound and video images. The MPEG standard has gained wide acceptance in satellite broadcasting, CD-ROM publishing, and DAB. The MPEG Standard is useful in a variety of products including digital compact cassette decoders and encoders, and minidisc decoders and encoders, for example. In addition, other audio standards, such as the Dolby AC-3 standard, involve the encoding and decoding of audio and video data transmitted in digital format.
The AC-3 standard has been adopted for use on laser disc, digital video disk (DVD), the US ATV system, and some emerging digital cable systems. The two standards potentially have a large overlap of application areas.
Both of the standards are capable of carrying up to five full channels plus one bass channel, referred to as “5.1 channels,” of audio data and incorporate a number of variants including sampling frequencies, bit rates, speaker configurations, and a variety of control features. However, the standards differ in their bit allocation algorithms, transform length, control feature sets, and syntax formats.
Both of the compression standards are based on psycho-acoustics of the human perception system. The input digital audio signals are split into frequency subbands using an analysis filter bank. The subband filter outputs are then downsampled and quantized using dynamic bit allocation in such a way that the quantization noise is masked by the sound and remains imperceptible. These quantized and coded samples are then packed into audio frames that conform to the respective standard's formatting requirements. For a 5.1 channel system, high quality audio can be obtained for compression ratio in the range of 10:1.
The transmission of compressed digital data uses a data stream that may be received and processed at rates up to 15 megabits per second or higher. Prior systems that have been used to implement the MPEG decompression operation and other digital compression and decompression operations have required expensive digital signal processors and extensive support memory. Other architectures have involved large amounts of dedicated circuitry that are not easily adapted to new digital data compression or decompression applications.
An object of the present invention is provide an improved apparatus and methods of processing MPEG, AC-3 or other streams of data.
Other objects and advantages will be apparent to those of ordinary skill in the art having reference to the following figures and specification.
SUMMARY OF THE INVENTION
In general, and in a form of the present invention a data processing device for processing a stream of data is provided which has software routines for managing an input buffer in response to breakpoint interrupts. A portion of memory is designated as in input buffer region for holding a portion of the input data stream and a second portion of memory is designated as a breakpoint queue to hold a sorted list of breakpoint addresses which point to selected locations in the input buffer. A software routine selects an address from the breakpoint queue and loads it into a breakpoint register which is then compared with each address presented to the memory. An interrupt request is generated when a match occurs. A processing unit responds to the interrupt by executing a software routine which interprets the reason for the interrupt and manages the input buffer accordingly. A breakpoint is used to indicate the end of the input buffer region. A breakpoint is used to delineate different types of data in the input buffer. A breakpoint is used to identify a CRC word. A breakpoint is used to detect buffer underflow and overflow.
In another form of the invention, a second breakpoint register and comparator is provided so that read and write breakpoints can be distinguished. A read pointer and a write pointer which point to the beginning of data and end of data in the input buffer are maintained in response to breakpoint interrupts.
Other embodiments of the present invention will be evident from the description and drawings.
REFERENCES:
Edition, Feb. 20, 1997 (ISO/IEC JTC1/SC29/WG11 N1519), Int'l Org. for Standardisation Coding of Moving Pictures and Audio.
Digital Audio Compression Standard(AC-3), Dec. 20, 1995, Advanced Television Systems Committee, ATSC Standard.
TI-17424A (S.N. 08/475,251), allowed,Integrated Audio Decoder System and Method of Operation.
TI-17600 (S.N. 08/054,127), allowed,System Decoder Circuit With Temporary Bit Storage and Method of Operation.
TI-24442P (S.N. 60/030,106), filed Provisionally Nov. 1, 1996.Integrated Audio/Video Decoder Circuitry.
Laczko, Sr. Frank L.
Li Stephen (Hsiao Yi)
Rowlands Jonathan
Brady III W. James
Harvey Minsun Oh
Laws Gerald E.
Telecky , Jr. Frederick J.
Texas Instruments Incorporated
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