Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2001-09-27
2004-05-04
Dorvil, Richemond (Department: 2617)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S222000, C704S225000, C704S200100
Reexamination Certificate
active
06732071
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to the field of signal processing. More specifically, the present invention relates to a method, apparatus, and system for efficient rate control in audio encoding.
BACKGROUND OF THE INVENTION
As technology continues to advance and the demand for video and audio signal processing continues to increase at a rapid rate, effective and efficient techniques for signal processing and data transmission have become more and more important in system design and implementation. Various standards or specifications for audio signal processing have been developed over the years to standardize and facilitate various coding schemes relating to audio signal processing. In particular, a group known as the Moving Pictures Expert Group (MPEG) was established to develop a standard or specification for the coded representation of moving pictures and associated audio stored on digital storage media. As a result, a standard known as the ISO/IEC 11172-3 (Part 3—Audio) CODING OF MOVING PICTURES AND ASSOCIATED AUDIO FOR DIGITAL STORAGE MEDIA AT UP TO ABOUT 1.5 MBITS/S (also referred to as the MPEG standard or MPEG specification herein), published August, 1993, was developed which standardizes various coding schemes for audio signals, e.g., MPEG-1 or MPEG-2 Layers I, II, and III. ISO stands for International Organization for Standardization and IEC stands for International Electrotechnical Commission, respectively. Generally, the MPEG audio specification does not standardize the encoder but rather the type of information that an encoder needs to produce and write to an MPEG compliant bitstream, as well as the way in which the decoder needs to parse, decompress, and resynthesize this information to regain the encoded audio signals. In particular, MPEG standard is developed for perceptual audio coding rather than lossless coding. In lossless coding, redundancy in the waveform is reduced to compress the sound signal and the decoded sound wave does not differ from the original sound wave. In contrast, in perceptual audio coding, the aim is not to regain the original signal exactly after encoding and decoding but rather to eliminate those parts of the audio signal that are irrelevant to the human ear (e.g., that are not heard).
An audio encoder typically includes a bit allocation module or unit (also called the bit allocator herein) whose role is to allocate more bits to those frequencies where quantization noise is audible to a listener and allocate fewer bits to those frequencies where quantization noise is masked and is inaudible to the listener. Also, the bit allocator needs to ensure that the total number of bits used for a specific audio block or frame does not exceed the maximum number of bits available as determined by the specified output bit rate. Currently, the methods for performing the bit allocation, as described in the MPEG standard includes two processing loops: (1) an outer or distortion control loop; and (2) an inner or rate control loop. One of the problems or disadvantages associated with the current methods described in the ISO/IEC 11272-3 MPEG standard is their inefficiency due to numerous iterations involved in determining or computing the optimum quantization parameters that will satisfy the rate criteria.
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Lopez-Estrada Alex A.
VanDeusen Mark P.
Blakely , Sokoloff, Taylor & Zafman LLP
Dorvil Richemond
Intel Corporation
Patel Kinari
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