Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion – With content reduction encoding
Reexamination Certificate
1998-11-17
2001-02-13
Hudspeth, David R. (Department: 2741)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
With content reduction encoding
C704S500000, C704S229000, C704S230000
Reexamination Certificate
active
06188987
ABSTRACT:
TECHNICAL FIELD
The present invention relates generally to audio signal processing in video/audio applications. More particularly, the present invention relates to encoding and decoding information conveyed in gaps or guard bands between frames of audio information that may be aligned with or, alternatively, embedded into frames of video information.
BACKGROUND ART
Several international standards have been developed which define various aspects of embedding digital audio information into frames of video information. For example, standard SMPTE 259M published by the Society of Motion Picture and Television Engineers (SMPTE) defines a Serial Digital Interface (SDI) in which up to four channels of digital audio information may be embedded into component and composite serial digital video signals. Standard SMPTE 272M provides a full definition of how digital audio information is to be embedded in ancillary data spaces within frames of the video information.
The serial transmission of digital audio information itself is the subject of various international standards. For example, standard AES3 (ANSI S4.40) published by the Audio Engineering Society (AES), defines serial transmission of two-channel digital audio represented in a linear pulse code modulation (PCM) form. According to this standard, PCM samples for two channels are interleaved and conveyed as pairs in two-sample frames.
A common activity in nearly all recording and broadcasting applications is editing or cutting embedded video/audio information streams and splicing the cut information streams to form a new single stream. Similar activities generate an information stream by merging multiple information streams or by switching between multiple streams. The video information is normally the primary synchronizing reference so that an edit or cut point is normally aligned with a video frame.
Standards such as AES11 define recommended practices for synchronizing digital audio equipment in studio operations. AES11 is directed toward controlling timing uncertainties caused by jitter or processing delays and provides for aligning video frame information with the two-sample frames of AES3 digital audio information streams. Equipment and methods that adhere to this standard can ensure that synchronized signals have the same number of frames over a given period of time and contain samples that have a common timing. Unfortunately, no standards or practices currently exist which define an alignment between video information and larger intervals of audio information. As a result, equipment from different manufacturers and even from the same manufacturer have variations in timing and in processing delays that introduce a significant amount of uncertainty in the relative alignment of audio and video information.
This uncertainty in alignment is of little consequence in applications that represent audio information in a manner such as that defined in the AES3 standard. Because edit points are constrained to occur between the two-sample frames of audio information, any uncertainty in video/audio alignment will not result in the loss of audio information. It will only affect the relative timing of sound and picture as presented to a person, which is unlikely to be discernable.
There is, however, a growing number of applications that use bit-rate-reduction encoding techniques to embed greater numbers of audio channels into a video/audio data stream. These encoding techniques are often applied to sample blocks of 128 or more audio samples to generate blocks of encoded information. These sample blocks typically represent audio information that spans an interval of 5 to 24 ms. Each block of encoded information generated by these encoding processes represents the smallest unit of information from which a reasonably accurate replica of a segment of the original audio information can be recovered. Split-band coding is one example of a bit-rate-reduction technique that reduces bit rates by applying psychoacoustic-based coding to frequency-subband representations of an audio signal. The frequency-subband representations may be generated by application of a plurality of bandpass filters or one or more transforms. For ease of discussion, these split-band coding techniques are described here in terms of applying a filterbank to generate subband signals.
The uncertainty in alignment mentioned above is significant in these block-coding applications because an edit point falling within the boundaries of an encoded block will result in part of that block being cut from the remaining signal. The partial loss of an encoded block will be manifested by a loss in the recovered signal for the entire block, which is for a duration typically of 5 ms or more. It is likely that such a loss would be discernable to the human auditory system.
This problem may be avoided by using a post-processing process in which a PCM representation of the original audio signals is recovered by applying a decoding process to the encoded audio, editing the recovered PCM representation as required, and generating a new encoded representation by applying an encoding processed to the edited PCM audio information. This solution is unattractive because of the additional costs and degradation in audio quality resulting from the decoding/re-encoding processes.
This problem may also be avoided by providing gaps or guard bands in the encoded audio information at locations of possible edit points. The guard bands have sufficient duration to span across the variations in alignment mentioned above so that an edit point will fall somewhere within a guard band. An edit point that falls within a guard band does not cause any loss of audio information. Unfortunately, the additional space or bandwidth required to convey the guard bands is essentially wasted because it is not used to convey any useful information.
DISCLOSURE OF INVENTION
It is an object of the present invention to utilize the information carrying capacity of guard bands used to separate frames of encoded audio information.
According to the teachings of one aspect of the present invention, a method encodes audio information and auxiliary information by receiving an audio signal conveying audio information, receiving a reference signal, receiving an auxiliary signal conveying auxiliary information, constructing auxiliary segments each comprising a block of auxiliary information and an error detection code, and generating an output signal conveying the audio information arranged in frames separated by gaps such that each gap is aligned with the reference signal and conveys a respective auxiliary segment, wherein the audio information in the frames of audio information and the auxiliary information in the segments of auxiliary information are constrained to avoid data patterns in a set of one or more forbidden data patterns.
According to the teachings of another aspect of the present invention, a method decodes audio information and auxiliary information by receiving an input signal conveying audio information arranged in frames separated by gaps and conveying auxiliary segments arranged in the gaps, processing the input signal to locate the auxiliary segments, for each respective auxiliary segment, extracting a block of auxiliary information and an error detection code, and evaluating the error detection code to validate contents of the block of auxiliary information.
The various features of the present invention and its preferred embodiments may be better understood by referring to the following discussion and the accompanying drawings in which like reference numerals refer to like elements in the several figures. The contents of the following discussion and the drawings are set forth as examples only and should not be understood to represent limitations upon the scope of the present invention.
Throughout this discussion, more particular mention is made of audio information that is encoded into blocks and conveyed in frames of one or more blocks that are aligned with video information. These references to audio and video information me
Chawan Vijay B
Dolby Laboratories Licensing Corporation
Gallagher & Lathrop
Hudspeth David R.
Lathrop David N.
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