Coded data generation or conversion – Digital code to digital code converters – Data rate conversion
Reexamination Certificate
1999-09-28
2001-08-21
Tokar, Michael (Department: 2819)
Coded data generation or conversion
Digital code to digital code converters
Data rate conversion
38, 38, C370S321000, C370S326000, C370S337000, C370S260000, C370S265000
Reexamination Certificate
active
06278387
ABSTRACT:
BACKGROUND
1. Technical Field
The present invention relates to the field of encoding and decoding of audio signals. More specifically, it relates to audio encoding and decoding systems (including MPEG-1 and MPEG-2 compliant systems) that enable variable playback of audio signals.
2. Description of Related Art
A conventional audio encoding system typically compresses an audio signal either to conserve storage space or prior to transmitting the audio signal. One method of compression involves the splitting of the audio signal into several frequency sub-bands before encoding (e.g., as utilized by motion picture expert group standards, MPEG-1 and MPEG-2 compliant encoding systems).
Conventional MEPG-1 and MPEG-2 compliant systems define several encoding schemes that utilize sub-band filtering for encoding audio-visual information. After encoding an audio signal using any one of these schemes, the encoded signal is either transmitted or stored for play back at some subsequent time. An audio decoder is then employed to decompress the encoded signal for play back.
When the encoded audio signal is played back at a normal rate using a conventional audio decoder system, the quality of the audio signal is relatively high. The user, however, may wish to increase or decrease the playback rate, e.g. at twice (2×) the normal speed. One example concerns the playback of video film for review where users wish to increase or decrease the rate of playback.
Conventional decoder systems are unable to playback audio signals at speeds other than normal. Further disadvantages of the related art will become apparent to one skilled in the art through comparison of the related art with the drawings and the remainder of the specification.
SUMMARY OF THE INVENTION
Various aspects of the present invention can be found in an audio codec that includes an encoder for encoding a first audio signal and a decoder for decoding a second audio signal. Also included is a rate adjust module, that permits variable playback of the second audio signal. While the first audio signal may be PCM samples stored on a storage media, the second audio signal may be a compressed bit stream received through a communication channel. Alternatively, the second audio is signal may be a compressed bit stream of the first audio signal.
The encoder includes an input filter bank that splits the first and second audio signals into a first, second, and up to thirty-two sub-band frequency signals, respectively, as specified under MPEG-1 and MPEG-2. The encoder further includes a psycho-acoustic model, a bit allocate circuitry, a formatter, and an output interface that outputs a compressed audio bit stream corresponding to the received PCM samples.
The decoder includes an input interface, an unformatter, an inverse bit allocate decode, and a time scaling module that time stretches received input samples within the time domain for each of the first and second frequency sub-bands to enable variable playback of the received (compressed) audio bit stream. The decoder further includes an output filter bank, and a digital to analog converter that converts the input samples to a corresponding analog signal.
In one embodiment, the time scaling module forms the input samples into an input frame and an output frame, overlaps the input and the output frames at a best averaging point, and averages the overlapped portions of the input and output frames at the best averaging point. Typically, the best average point is within a search range that has a minimum and a maximum value (in samples). The minimum and the maximum value each sub-band, is predetermined based on the sampling frequency of the audio samples. The time scaling module time may either compress or expand the audio samples for playback.
Aspects of the present invention may also be found in a method utilized by a time scaling system to manipulate samples of an audio signal. The method includes receiving the audio samples having a first and a second sub-band frequency, forming an input and a first output frame using the audio samples, computing a best averaging point within a search range for overlapping the input and the first output frame, overlapping the input frame and the first output frame at the averaging point by fading in and fading out the audio samples, and averaging the input and the first output frame at the best averaging point to form a second output frame. In utilizing audio samples to form an input and an output frame, the number of audio samples within an input frame may be determined. The number of audio samples within an input frame may be fixed or user-selectable.
Other aspects of the present invention will become apparent with further reference to the drawings and specification which follow.
REFERENCES:
patent: 4862168 (1989-08-01), Beard
patent: 4933675 (1990-06-01), Beard
patent: 5451954 (1995-09-01), Davis et al.
patent: 5712635 (1998-01-01), Wilson et al.
patent: 5786778 (1998-07-01), Adam et al.
patent: 5896099 (1999-04-01), Yamauchi
Akin Gump Strauss Hauer & Feld L.L.P.
Conexant Systems Inc.
Mai Lam T.
Tokar Michael
Tyson, Jr. H. Shannon
LandOfFree
Audio encoder and decoder utilizing time scaling for... does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with Audio encoder and decoder utilizing time scaling for..., we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Audio encoder and decoder utilizing time scaling for... will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-2451852