Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
• • G—PHYSICS
  • G11—INFORMATION STORAGE
  • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
  • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
  • G11B20/10—Digital recording or reproducing
  • G11B20/18—Error detection or correction; Testing, e.g. of drop-outs
  • G11B20/1876—Interpolating methods
• • G—PHYSICS
  • G11—INFORMATION STORAGE
  • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
  • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
  • G11B20/24—Signal processing not specific to the method of recording or reproducing; Circuits therefor for reducing noise

Definitions

• the invention relates to a method according to the preambles of the independent claims set forth herein for the reconstruction of an audio signal and a computer software product.
• the method according to the invention for reconstruction of audio signal is especially suitable for correcting errors of short duration in an audio signal, e.g. correcting errors in records when an LP record is stored in digital form.
• a typical problem that occurs when an LP record is stored in digital form is the poor quality of the record as a result of scratches or other errors in it, whereby a mechanical error on the surface of the LP record, for example, causes an abrupt movement of the stylus of the record player.
• the simplest way of correcting the error is editing, that is just cutting off the point of error.
• the empty point is removed by continuing the signal from the starting point of the error directly with a signal after the error.
• the erroneous signal can also be corrected by filtering off signals that are either above or below a predetermined frequency range. However, this will also deteriorate the quality of faultless signal areas.
• the patent publication EP 0336685 discloses a method for correcting the point of error in the signal by using an autoregressive model (AR model).
• AR model is used over the whole area of data where the error is, whereby the same group of equations includes simultaneously the known data before and after the error data as well as the actual error data. Because the error data and the AR coefficients a ⁇ - are in the same group of equations, they can interact with each other, whereby the solution is instable.
• the method disclosed in the publication also has the drawback that the solution requires large groups of equations, which makes calculation time-consuming. Another shortcoming of the solution is the fact that it is iterative.
• a Japanese patent publication JP 63086163 discloses an iterative method based on linear prediction for the restoration of a missing signal.
• the impuls response of a faultless signal is not used in the method, as a result of which the error contained by the predicted signals increases exponentially the longer the sequence of missing signals is.
• an attempt was made to remove this error by comparing the last predicted signal to the signal of the first area with errors, whereby it was possible to estimate the size of the error.
• the signal derived by prediction is accepted when the detected error does not exceed a predetermined threshold value. If the error exceeds the threshold value, the erroneous value is replaced by a value obtained by interpolation.
• S. Montresor et al. disclose a method in their publication [3] wherein impuls noise occurring in gramophone records is corrected by using an autoregressive model.
• the calculation of the coefficients used in the AR model described in the specification is very complicated, which reduces the usability of the model.
• S.V. Vaseghi and R. Frayling-Cork disclose a method in their publication [4] for eliminating impulsive errors in the signals of LP records.
• the impulsive error is treated by creating an error model, which consists of the mean value of a number of similar real errors.
• the oscillating portion of an impulsive error in the signal can be cancelled by subtraction by means of the impulsive error model.
• subtraction does not help, because the peak has caused irrecoverable damage to the signal.
• the erroneous data in the area damaged by the peak are corrected by the iterative interpolation method described in the publication [5]. Nevertheless, erroneous data are used in this method for calculating the corrective data.
• known methods also have the problem that the length of the erroneous area is limited.
• the known methods can be used to correct an erroneous area, which contains some tens, or at the most a little over one hundred data points.
• the known methods have the drawback of the high calculation power required, because methods that include calculation are based on very complicated mathematical models, and they are mainly iterative.
• the audio signal is typically reconstructed so that the impulse responses of the signals are at first calculated from the faultless area before and after the area with errors.
• the signal is extrapolated forward from the beginning of the area with errors by using an impulse response calculated from the area before the area with errors, and backward from the end of the area with errors by using an impulse response calculated from the area after the area with errors.
• the erroneous signal is replaced by a linear combination of the signals obtained by extrapolation.
• the signal is extrapolated from both directions of the erroneous area and the erroneous signal is replaced by the linear combination of the signals obtained by extrapolation, the formation of discontinuities at the ends of the area with errors is prevented.
• the frequency variations in the erroneous area are naturally changed by sliding due to the weighting function.
• the extrapolated signal is obtained as a convolution of a signal preceding the area with errors and its impulse response.
• the method according to the invention applies direct extrapolation, in which a new signal point obtained by the extrapolation of one point is used for the calculation of the next missing signal point.
• the method according to the invention can be reliably and accurately be used to replace even a long erroneous signal by a signal obtained by extrapolation.
• the number of impulse response points used varies from 100 to 5000, typically between 500 and 2000 and is advantageously 1000 impulse response points.
• the extrapolation is based on sufficient information about the strongest frequencies contained by the signal and their amplitude behaviour.
• the method is used for saving LP records in digital form.
• errors in the LP record can be corrected in connection with digital saving in a quick and simple manner.
• the method according to the invention can also be used to correct errors contained by a speech signal, such as a speech signal transmitted by mobile stations and radios.
• the method according to the invention is typically implemented with a computer software product, which can be directly loaded to the central memory of the computer or with a computer software product saved in a computer- readable medium, which computer software product contains program code elements for implementing the method according to the invention, when said computer software product is run in a computer.
• a computer software product means independent computer software or a part of computer software, which may comprise one or more program code elements.
• a program code element means an element that consists of one or more computer-readable instructions.
• a computer-readable medium means all mediums in which information, such as instructions, commands, instruction or command sequences or the like, which are readable by a computer, can be saved permanently or temporarily. Mediums like this include e.g. fixed disks, mass storages, cache memories, diskettes and CD-ROM discs.
• a typical computer software product as mentioned above, which is saved in a computer-readable medium, comprises at least the following program code elements:
• the method according to the invention for the reconstruction of an audio signal can be used in a computer or as part of a device, which includes means for saving the computer software product so that by means of it the method can be automated by using automatic data processing.
• An advantageous computer software product also comprises a program code element for making the computer recognize an area or areas with errors in the signal.
• the recognition may be based e.g. on a procedure in which the program code element searches for areas of the audio ⁇ signal in which the absolute value of the fourth derivate exceeds the squared mean value of the fourth derivate multiplied by a certain coefficient, and the program code element defines the areas found as areas with errors.
• the search for erroneous areas can be automated and thus the reconstruction of the audio signal can be made a faster and more accurate process.
• the greatest advantage of the method according to the invention is the fact that it enables the reconstruction of erroneous areas of thousands of data points reliably, quickly and accurately.
• the invention has the advantage that the method requires less calculation capacity than many prior art methods, whereby the invention can be implemented with relatively simple means. Thus the manufacturing costs of apparatus for the method do not become unreasonably high.
• Figure 1 illustrates the recognition of errors in an audio signal
• Figure 2 is a schematic diagram of correcting an error in an audio signal by linear prediction, in which the erroneus signal is at the top and the corrected signal is at the bottom,
• Figure 3 is a schematic diagram of the linear weighting function
• Figure 4 is a schematic diagram of a general weighting function
• Figs. 5a-5d show examples of the reconstruction of an audio signal by means of a computer sofware product, utilizing the method according to the invention.
• Fig. 1 shows as an example the part of the original analogue recording, which is A/D converted in the memory of the computer as the audio signal 1.
• a mechanical error on the surface of an LP record causes an abrupt movement of the stylus of the record player.
• this causes a quick change of amplitude values in time.
• derivative d j is obtained according to the definition of the derivative by calculating the difference of two consecutive signal data and dividing it by the distance between them
• the error searching method can also be applied so that the method searches for such areas 3 of the audio signal 1, in which the absolute value 2 of the fourth derivative exceeds the squared mean value 4 of the fourth derivative multiplied by a certain coefficient.
• the correction of a single error by linear prediction from an audio signal 21 in digital form is done by calculating at first the impulse response h from the faultless area 23 before the area with errors 22 and the impulse response h ' from the faultless area after the area with errors 22.
• the signal 21 is extrapolated by using the impulse responses calculated above, i.e. the signal 21 is continued by linear prediction from the beginning of the area with errors 22 to the end by using the impulse response h, whereby a signal s ' predicted forward is obtained, and from the end of the area with errors 22 to the beginning by using the impulse response h', whereby a signal predicted backward s" is obtained.
• the erroneous signal is replaced by the weighted average 25 of the signals s ' and s ' ' predicted forward and backward.
• t a is the starting point of the area with errors and ty is the ending point of the area with errors, as shown in Fig. 3.
• a linear weighting function is not optimal, because the accuracy of the prediction falls exponentially in proportion to the length of the prediction.
• Another extreme type of weighting function would be a function where the forward predicted signal s would be used only up to the mid-point of the area with errors 22, and after that only the backward predicted signal s' .
• the weighting function/? ⁇ would then be a step function, which would change from 1 to 0 in the middle of the area with errors. However, this might cause a point of discontinuity in the middle of the area with errors 22, and that would again be a new error in the signal.
• the most advantageous weighting function found between the above mentioned cases is shown in Fig. 3.
• the weighting function can be written in a general form
• Figs. 5a-5d show examples of the application of a computer software product in the reconstruction of an audio signal. According to what is shown in Fig. 5a, an error area 51 is detected by the eye in a stereo signal saved from an LP record.
• this area is selected for reconstruction by marking it with the cursor.
• the Burg method is selected as the calculation method for impulse responses, and the lengths of the impulse responses and the amount of faultless data used for the calculation are selected.
• the program calculates the reconstructive signal data and replaces the erroneous data by corrected data. The calculation and replacement operations are performed separately on both channels.
• Fig. 5d shows a reconstructed stereo signal.
• s " j is the backward extrapolated signal data.
• s " j is the backward extrapolated signal data.
• the following data can be calculated by using equation 10 of 11 again.
• the signal can be extrapolated without limits.
• the phase and amplitude information for the forward extrapolated signal comes from the faultless signal before the error, and the frequency and amplitude changing information comes from the impulse response obtained from the area preceding the error.
• the phase and amplitude information and the frequency and amplitude changing information of the backward extrapolated signal is obtained on the basis of the area after the error.
• the impulse responses h k and h ' k used in the calculation can be calculated for the known signal in many different ways.
• S j S j in the equations 10 and 11.
• the known signal S j is calculated by means of M+ 1 previously known signal data. If there are 2M + 1 known data, it is possible to solve h ' k and h k from the equations 10 and 11. Extra known data can also be included, whereby the matching of the smallest square sum is used in the calculation of the impulse response of M.
• the method described above is a so-called matrix method.
• Another method for the calculation of the impulse response is the Burg method [6, 7], which is in practice very suitable for audio signals because of efficient calculation and a good stability.

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• Engineering & Computer Science (AREA)
• Signal Processing (AREA)
• Computational Linguistics (AREA)
• Health & Medical Sciences (AREA)
• Audiology, Speech & Language Pathology (AREA)
• Human Computer Interaction (AREA)
• Physics & Mathematics (AREA)
• Acoustics & Sound (AREA)
• Multimedia (AREA)
• Complex Calculations (AREA)
• Signal Processing For Digital Recording And Reproducing (AREA)

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• 2000
  • 2000-03-07 FI FI20000523A patent/FI117147B/sv not_active IP Right Cessation
• 2001
  • 2001-03-05 WO PCT/FI2001/000213 patent/WO2001067451A1/en not_active Application Discontinuation
  • 2001-03-05 EP EP01917133A patent/EP1277208A1/en not_active Withdrawn
  • 2001-03-05 AU AU2001244235A patent/AU2001244235A1/en not_active Abandoned

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