Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
1999-01-11
2002-01-08
Korzuch, William (Department: 2641)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C380S201000, C705S058000
Reexamination Certificate
active
06338037
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to labelled audio signals to enable subsequent identification, and in particular to the decoding of code labels embedded within audio signals.
Code labelling of audio and/or video sound track recordings commonly occurs to indicate the origins of the recordings, or the owner of the copyright in the recordings, or both. The labelling may also provide information as to payment of copyright royalties due.
BACKGROUND ART
Our earlier patent EP-B-0245037 and our International Patent Application PCT/GB95/03035 (corresponding to Patent EP 081855) both relate to apparatus for the labelling of an audio signal, comprising two notch filters to form notches at specified first and second frequencies; code generating means to produce a code label signal for the audio signal, having a plurality of bits, a “one” bit value represented by a burst at the first frequency and a “zero” bit value being represented by a burst at the second frequency, the code being inserted into the audio signal at the notch frequencies at a predetermined amplitude level relative to the audio signal. When for example the audio signal is broadcast, and a received signal is decoded, complementary band pass filters at the first and second frequencies are operative to extract the frequency bursts, and the signal is then decoded.
Our patent EP 0366 381 proposes a signal encoding means for embedding code signals in an audio signal recorded on an audio track. In order to make it more difficult for a defrauder to locate the code signals, the system proposed to eliminate the creation of a predictable sequence of notch frequencies where the code signal is embedded, by generating a sequence of notch frequencies which varies throughout the audio sound track recording. The system proposes the use of wide pass band filters to accommodate fluctuations in the speed of the recording mechanism.
Usually, the code label signal consists of two digital words, each word including an initial portion comprising a simultaneous burst of both frequencies. A data portion then follows comprising bursts of either the first or the second frequency to represent a “1” bit or a “0” bit. More than one digital word may be necessary on account of the amount of data to be inserted if the International Standard Recording Code (ISRC) is to be represented.
It has been found that the decoding of such a coded audio signal is sensitive, where say the audio signal results from the playing of a tape or a CD, to speed variations in the tape or CD playing mechanism, resulting in variations in the values of the first and second code frequencies, since the code label frequencies will also be recorded on the CD or tape. Significant losses of accurately decoded code signals may result for speed changes at or beyond ±0.5%. This figure of 0.5% depends on the dimensions of the coding notch and decoding band pass filter widths in use. The problem of speed variation might in principle be resolved by the use of wider filters having less speed sensitivity as proposed in our European Patent 0366 381. But wider filters create more risk of the code insertion process producing audible artefacts. Although there are different causes for speed variation, for example wow and flutter type speed variations, the present invention is particularly concerned with constant speed errors such as may occur with a variable speed CD player when a radio station speeds up the CD by a percent or two to squeeze the record in before the end of a programme.
SUMMARY OF THE INVENTION
In order to overcome problems of variations in the frequencies at which code labels are present in audio signals, the present invention contemplates a decoder for retrieving code label signals having a respective decoder pass band filter arrangement for each of the code frequencies, the pass band filter arrangement having a center frequency which is varied or selected from a number of possible values such that if the code frequency varies from the nominal value of the frequency, encoded code label signals can still accurately be received.
The present invention provides in one aspect apparatus for decoding code label signals encoded into audio signals, the code label signals comprising bursts of at least first and second specified frequencies, and the apparatus including a plurality of pass band filter arrangements, one for each of the specified frequencies, each pass band filter arrangement having a center frequency which can be varied or selected from a plurality of possible values such that if the respective code frequency varies from its nominal value, encoded code label signals can still accurately be received and decoded.
In one embodiment, the pass band filter arrangement comprises a tracking or adaptive filter whose center frequency is variable in dependence on the quality of the received signals, so that the filter can accurately follow the incoming code frequency. Usually, the tracking filters for the respective code frequencies will be ganged together so that they vary by the same amount and in the same direction, since frequency changes due to speed variations will affect all the code frequencies equally. In some circumstances however, it may be desirable to have independent tracking filters, where the code frequencies are likely to change by different amounts, for example as a result of changes introduced by signal processing.
However a tracking filter embodiment may not be suitable for short code signals, since there will be inevitably be an initial lock on time which may amount to a significant part of the code label duration. It is therefore presently preferred to employ pass band filter arrangement with a plurality of pass band filters with slightly different pass band frequencies values in a range closely surrounding the nominal audio frequency and arrange simultaneously to receive the audio signal. A suitable means is provided to select the best quality output from the filters. Thus effectively the value of the center frequency is selected from a plurality of values in dependence on the quality of received data.
It would be possible to multiplex the outputs of the filters to a single decoder, followed by a data selector for selecting the best quality data. Alternatively, the data selector would precede the decoder. However both of these alternatives would place heavy demands on the hardware, and it is therefore preferred to employ a plurality of decoders, comprising pass band filters and signal analysis means, arranged in parallel and outputting to a single data selector operative to select the best quality data. In a preferred example, four decoders may be employed, each decoder including a plurality of pass band filters for the respective code frequencies, each filter deviating by the same amount from the nominal code frequency.
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Johnson Nigel
Todd Martin Peter
Central Research Laboratories Limited
Korzuch William
Lerner Martin
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