WO2001009889A1 - Dispositif et procede de marquage de donnees audio - Google Patents

Dispositif et procede de marquage de donnees audio Download PDF

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Publication number
WO2001009889A1
WO2001009889A1 PCT/IB2000/001138 IB0001138W WO0109889A1 WO 2001009889 A1 WO2001009889 A1 WO 2001009889A1 IB 0001138 W IB0001138 W IB 0001138W WO 0109889 A1 WO0109889 A1 WO 0109889A1
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WO
WIPO (PCT)
Prior art keywords
spectral
audio data
representation
information
spectral components
Prior art date
Application number
PCT/IB2000/001138
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English (en)
Inventor
Patrick O'neal Nunally
Original Assignee
Global Intertech Marketing Limited
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Global Intertech Marketing Limited filed Critical Global Intertech Marketing Limited
Priority to AU63110/00A priority Critical patent/AU6311000A/en
Publication of WO2001009889A1 publication Critical patent/WO2001009889A1/fr

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Classifications

    • 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
    • G11B20/00086—Circuits for prevention of unauthorised reproduction or copying, e.g. piracy
    • G11B20/00884—Circuits for prevention of unauthorised reproduction or copying, e.g. piracy involving a watermark, i.e. a barely perceptible transformation of the original data which can nevertheless be recognised by an algorithm
    • G11B20/00891—Circuits for prevention of unauthorised reproduction or copying, e.g. piracy involving a watermark, i.e. a barely perceptible transformation of the original data which can nevertheless be recognised by an algorithm embedded in audio data
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
    • G10L19/018—Audio watermarking, i.e. embedding inaudible data in the audio signal
    • 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
    • G11B20/00086—Circuits for prevention of unauthorised reproduction or copying, e.g. piracy
    • 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
    • G11B20/10527—Audio or video recording; Data buffering arrangements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/66—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission
    • H04B1/667—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission using a division in frequency subbands

Definitions

  • the invention concerns the marking of audio data in order to authenticate it or restrict its use. More particularly, the invention concerns marking of audio data by imposition of a code on the spectral content of the audio data.
  • Audio information is rendered in the form of audio data, that is audio information in a form on which computer programs operate.
  • Audio information is, generally, that information that may be perceived by the human auditory system.
  • Such information is typically in the form of sound that is composed of signals in the frequency range of about 15 hertz to 20,000 hertz.
  • Audio data therefore, is sound that has been converted into a format that can be operated on by computers.
  • Audio data is available and may be apprehended from broadcast and cable media, over networks (such as the internet), and from digital storage media such as compact disks (CDs) and digital audio tapes (DATs), and equivalents.
  • the invention provides a technological solution to this problem by imposing on the spectral content of audio data information known and accessible only to the owner of the audio information and authorized representatives.
  • mark coding or “marking”.
  • the solution is embodied in a method and system in which a pseudo random sequence of numbers or indexes is generated in response to the audio information and to private information.
  • the pseudo random sequence of numbers or indexes points to specific spectral components of the audio information which are evaluated for marking with marking information. Marking is accomplished by altering the spectral content of the audio information in such a way as to be virtually inaudible.
  • the encoding of audio data with marking information adjusts the spectral distribution of energy in the audio information, without adding new frequency components or deleting existing ones. Consequently, the spectral population of the audio information does not change with mark coding.
  • decoding may be conducted using the spectral population of the mark coded audio information.
  • mark coding can be robust enough to withstand a number of quantizations, or it may be so fragile as to be lost with the first quantization following encoding.
  • the mark coding can combine robust and fragile components.
  • Figure 1 is a block diagram showing a prior art system for marking an image.
  • Figure 2 is a block diagram illustrating the architecture of a system according to the invention.
  • Figures 3A-3C are spectral diagrams illustrating how the spectral content of audio information is coded according to the invention.
  • Figure 4 is a flow diagram illustrating the operation of the system of Figure 1 in coding audio information with marking information.
  • Figure 5 is a block diagram illustrating incorporation of the invention into an audio encoder.
  • Figure 1 illustrates a prior art system for identifying an image based upon information in the image that is obtained, but distinct, from the contents of the image itself.
  • the information may either be derived from the image or entered into the image.
  • the system of Figure 1 derives, creates, or otherwise generates a digital mark that may compared with the contents of an image 10 for the purpose of identifying, authenticating, or otherwise validating the image 10.
  • the mark is derived by processing private information 11 in response to the contents of the image 10.
  • the private information may comprise any type of information in any form that can be transformed into a digital representation and that is private to a person, enterprise, or a machine having some relationship to the contents of the image 10.
  • the private information may comprise, for example, a private number set or sequence such as social security number, a driver's license number, a DNA sequence, a telephone number, a private character set or sequence, a private alphanumeric set or sequence, a private graphic, a private image, a private document, or a private code. It is necessary that the private information 11 be repeatable in the sense that, from one operation to another, the private information will not change.
  • the private information 11 is input to a digitizing process 12 which reduces the private information to a digital electronic form having a predetermined size.
  • the digital form of the private information is represented by a two-dimensional array 14 of ones and zeros that would reside, for example, in the memory of a computer.
  • the two-dimensional array 14, may of course be assembled into a unique uni-dimensional multi-bit vector by scanning it in some regular fashion.
  • the private information has been rendered in a repeatable manner into a digital object that may be processed in response to the contents of the image 10.
  • An electronic image 16 of the image 10 is obtained by conventional means.
  • the electronic image 16 may be a digital image of the image 10.
  • the digitizing element 18, for example, may implement the well-known ISO 1098 algorithm, with the product being an image feature such as spectral content that may be represented by a rectangular array 19 of samples, each sample being embodied in the digital number.
  • the private information in the digital array 14 is scrambled in response to the digitized image contents produced at 18, with the scrambling being done in a linear feedback shift register (LFSR) 20 that is seeded by a lxn digital number embodying the digitized private information available in the array 14.
  • the LFSR 20 is a shift register of n-stages whose operation is controlled by a sequence of clock pulses provided from a clock generator 22.
  • the operation of the LFSR may be understood with reference the Sklar's DIGITAL COMMUNICATIONS: Fundamentals And Applications, (Prentiss-Hall 1988), pp. 546-549.
  • the LFSR 20 scrambles the digital number embodying the private information in a series of shifts, each shift occuring in response to a clock pulse produced by the clock generator 22.
  • the clock generator 22 operates in response to the array of samples produced by the element 18.
  • an END OF ARRAY signal is produced that disables the clock generator 22 and unloads the contents of the LFSR 20.
  • the operation of the LFSR 20 will have scrambled the digital number that embodies the private information in a pseudo-random manner. This process is also referred to as "randomization” or "pseudo-randomization".
  • the LFSR contents are arranged into an array 24 of ones and zeros in a two-dimensional matrix.
  • This two-tone pixel array is presented at 26. It is possible to reduce the image of the pixel array 26 and use it as a mark 28 to apprehend or enter information into the contents of the image 10. This is done by correlating the mark with the image contents. The result is a correlation pattern. Either a code is derived by the correlation pattern, or a code is imposed by adjusting the correlation pattern. In the mode where the correlation pattern is adjusted, care is taken to make any change in the spectral content of the image virtually imperceptible to the eye.
  • a random or pseudo-random process is used to index into the spectral content of audio data in order to mark the audio data by encoding the data with information that is related to an interest in the program information embodied in the audio data.
  • information may, for example, signify an owner, a licensee, a distributor, a marketer, or any person or enterprise having an interest in the program information.
  • the marking information is encoded in the audio data in such a way as to make it virtually imperceptible to the human auditory system, yet tractable to a decoder. This process is referred to as "mark coding” or, simply, "encoding”.
  • FIG. 2 illustrates an exemplary system with which the invention may be practiced.
  • an audio program source 200 provides on 201 audio data embodying audio information program material.
  • the form of the information at 201 is presumed to be compressed, formatted information that is referred to hereinafter as "audio data".
  • the audio data at 201 may be generated using any of a number of audio compression algorithms and formats. Examples of such include AC3 (Dolby), Pro Logic, THX, MPG, and other equivalent schemes.
  • AC3 Dolby
  • the audio data at 201 is the result of spectral decomposition of audio information, followed by processing, quantization, and coding and formatting for transmission. Because the operations and functions of the invention preferably are directed to a spectral representation of the audio data to be encoded, the audio data at 201 is deformatted and expanded to provide a digital representation ("digital audio") of the audio information; this is accomplished at step 202.
  • This partial inversion of the MPEG process advantageously provides the digital representation in a sequence of frames. The frames are clearly marked in the MPEG scheme. (As mentioned above, the MPEG scheme is merely an example.)
  • block 202 of Figure 2 will provide a partition of digitized audio with a partition mark provided to indicate the boundary between adjacent partitions.
  • a partition is a frame and the partition mark indicates the boundary between adjacent frames.
  • a transform function at 208 transforms each unit of digitized audio into a spectral content signal that may be represented by a magnitude vs. frequency plot (also called a "spectral representation") having, for example, 1024 frequency bins.
  • the transformed unit of digitized audio is evaluated by a threshold function at 210 for the purpose of determining whether the unit can be encoded. Since, as will become clearer as this description progresses, encoding involves altering the distribution of energy in a transformed unit of audio data, the thresholding component 210 evaluates the distribution of energy among the spectral components of the unit. In this regard, the threshold component 210 identifies units with high spectral density.
  • the term "high spectral density” means that a spectral representation has active frequency components in no less that thirty percent of the frequency bins defined for the representation. Thus, for a 1024 bin fast fourier transform (FFT), for example, 307 bins would have to be active. Active frequency components are those exhibiting a signal magnitude above some noise floor. Thus, to determine which bins are "active", an average system noise magnitude is subtracted from the magnitudes of all bins in a sample. Once system noise is removed, the mean magnitude is calculated. If this mean falls above a predefined threshold, the frame is ready for coding, if below, the frame is not encoded.
  • FFT fast fourier transform
  • the threshold component 210 enables a clock generator 212 that generates one clock pulse for each frequency bin in the spectral representation of the current unit of digitized audio, with the condition that the magnitude of the frequency or frequencies in the bin exceed a minimum level.
  • the clock pulses produced by the clock generator 212 operate a linear feedback shift register (LFSR) 214 seeded with a digital representation of private information 215.
  • LFSR 214 contains a digital word from which a number can be extracted by, for example, truncation, modulo- arithmetic, and so on. The number obtained from the LFSR indexes to a bin in the spectral representation of the current unit of digitized audio.
  • the bin that is indexed by the contents of the LFSR 214 is inspected by a mark coding component 216 for encoding according to the invention.
  • a buffer 218 accumulates the succession of transformed units of digitized audio, both coded and uncoded, in the sequence that they are transformed at 208, feeding them in sequence to a inverse transform component 219.
  • the output of the inverse transform unit 219 is fed to a compression unit 220 that renders the now-encoded audio data into the form and format that the audio data had at 201.
  • Figure 3 A is a spectral plot representing data output by the transform component 208 that drives the clock generator 212 and that is buffered at 218.
  • Figure 3 A shows a plurality of frequency bins 1, 2, 3, . . ., j-1, j, j+1, . . . .
  • the clock generator 212 thresholds the magnitudes of the frequency bins, in sequence, comparing their contents against an established counting threshold magnitude 300.
  • the amount by which any bin's magnitude must exceed the magnitude level 300 provides a magnitude margin that will guard against loss caused by subsequent processing that involves quantization as would be required, for example, by decoding and also by reencoding following a decoding.
  • a binary bit can be encoded in the unit of digitized audio that has just been transformed by considering the immediate neighbors of bin j, that is bin j-1 and bin j+1.
  • a zero could be encoded according to the invention by making the magnitude of bin j exceed the magnitude of each of its immediate neighbors by a predetermined amount. If this were already the case, no further processing would be required to encode a zero. Presume, however, that the magnitude of the contents of bin j is less than the magnitude of the contents of each of its neighboring bins, as illustrated in Figure 3A.
  • the mark coding component 216 would redistribute the spectral energy between the three bins to achieve the results illustrated in Figure 3B. That is, a portion of the magnitude in bin j-1 would be deleted from that bin and added to the magnitude in bin j; similarly, a portion of the magnitude of the contents of bin j+1 would be subtracted from that bin and added to the contents of bin j.
  • Figure 3B This illustrates one way in which one binary state (in this case, a zero) can be encoded into the audio data by redistribution of the spectral energy in the audio information.
  • Figure 3C illustrates how another binary state (in this case a one) can be the spectral energy distribution of audio data.
  • the contents of bin k initially have a magnitude 305, while bins k-1 and k+1 have magnitudes (307 and 309, respectively) that are less than the magnitude of bin k.
  • a one is encoded by subtracting a portion of the magnitude of bin k and adding that portion to the contents of bin k-1 and subtracting another portion of the contents of bin k and adding that portion to contents of bin k+1.
  • the coding does not have to be binary, nor does it necessarily require three bins or even adjacent bins although adjacency greatly assists in making the marking imperceptible.
  • the invention does contemplate that the coding will be accomplished by redistribution of the spectral content of audio data.
  • the invention further contemplates that such spectral redistribution will take into account the counting threshold magnitude 300, taking care that the spectral redistribution will not change the number of bins causing clock pulses; this ensures that encoded bins can be identified and decoded.
  • Figure 4 illustrates a method for mark coding according to the invention and may be understood with reference to Figure 2.
  • the mark coding process starts at 400 with the presumption that decoded and decompressed audio data is fed in a sequence of defined and/or definable units to the transform component 208. Each unit of digitized audio is received and transformed by the process 208.
  • a decision 404 determines whether the spectral distribution indicates that the transformed unit has high spectral density. If not, the negative exit is taken and conventional buffer control is accessed at 405 to ensure proper operation of the buffer 218. If the transformed unit has high spectral density, the positive exit is taken from the decision 404 and the first symbol of a code symbol set 408 is encoded as pattern of ones and zeros ("bits"). A first location for the first bit is determined in step 410 by clocking the LFSR 214.
  • the first location is indexed by the contents of the clocked LFSR; this location is one of the bins in the last-transformed unit of digitized audio.
  • a bit is encoded at this location in step 412. For so long as bits remain to be encoded for the current symbol, the negative exit will be taken from decision 414, the next location accessed by cycling the LFSR 214 in step 410 and the next bit will be encoded in step 412.
  • the positive exit is taken from decision 414.
  • decision 418 if more symbols remain to be encoded, the positive exit is taken, the next symbol is obtained in step 407 and is encoded as just described. When all symbols have been encoded, the negative exit is taken from decision 418.
  • the buffer 218 is processed in step 405, the next unit of digitized audio is obtained and transformed, and the just-described process is repeated.
  • the mark coding process illustrated in Figure 4 can repeatedly encode a symbol sequence that forms the basis of marking information. Such information can be replicated throughout the audio program information, as deemed necessary for any particular application.
  • This procedure may be employed to mark audio data at any one of a number of points in an audio information processing and distribution configuration. In this regard, it may be used where audio data is introduced or staged through nodes in a communication system such as the internet. It can also be used to mark audio data that is being processed for storage on readable media such as CDs.
  • the architecture of Figure 2 can be adapted for an encoding appliance that receives analog audio and converts it into an encoded audio data format.
  • the procedure of Figure 4 can, of course, be implemented in a software program and executed in general purpose digital computer; alternatively, the architecture of Figure 2 and procedure of Figure 4 can be implemented in application specific integrated circuitry (ASIC).
  • ASIC application specific integrated circuitry
  • bits can be robustly encoded so as to mark audio information in a way that is tractable to a decoder after marked audio information has been subjected to quantization after mark coding.
  • Fragile mark coding limits the amount of energy redistributed between adjacent bins to a level that would result in loss of encoded bits after subsequent quantization; for example, with a known level of noise introduced by a quantization procedure utilized in a known compression algorithm, those skilled in the art will be able to encode bits according to this invention by redistributing an amount of energy that is small enough to result in a statistically significant loss of bits after marked audio data has been quantized subsequent to marking.
  • Decoding of marking information inserted into audio information presumes that the marking information is embedded in spectral energy by redistributing the energy contents of adjacent frequency bins to implement a binary code, for example, according to the encoding illustrated in Figures 3B and 3C. Extraction of the marking information requires the detection of mark coded bins by indexing using an LFSR as illustrated in Figure 2. Since the mark coding procedure does not alter the spectral population of a transformed unit of digitized audio, provision of the same private information used to mark code the audio information will result in production of numbers or indexes by the LFSR in the manner described above; in decoding, of course, the indexes will point to coding locations. Decoding will require detection of bit states, accumulation of bits, and identification of symbols. Of course, error correction codes may be employed with this invention.
  • FIG. 5 A particularly efficacious embodiment of the invention is illustrated in Figure 5, where an audio encoder 500 receives an unencoded analog audio signal from a source 502.
  • the analog signal is initially framed at 510.
  • Each frame is transformed (by FFT, DFT, or any equivalent) at 511 and then quantized at 512.
  • the output of the quantizer would be fed to a data coding and formatting unit 513.
  • the system architecture of Figure 2 can be adapted for the audio encoder 500.
  • This adaptation which is presented in Figure 5, assumes that audio data has already been transformed into its spectral components.
  • the output of the quantizer 512 is a series of spectral representations, each ready for mark coding according to the invention.
  • a mark coding apparatus 522 that operates according to the invention includes threshold, clock generation, LFSR, and mark coding components.
  • a seed memory or register 520 stores private information for the LFSR.
  • a buffer 523 stages transformed frames for processing by the marking apparatus 522 and then coding and formatting by data coding and formatting unit 513.

Abstract

L'invention concerne des données audio marquées de telle sorte que ce marquage soit inaudible pour le système auditif humain. Ce procédé de marquage consiste en l'indexation des données dans des représentations spectrales. L'indexation identifie les composants spectraux dont les magnitudes sont redistribuées afin de coder de manière binaire, ou encore, d'insérer des informations de marquage dans les informations audio.
PCT/IB2000/001138 1999-07-30 2000-07-24 Dispositif et procede de marquage de donnees audio WO2001009889A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU63110/00A AU6311000A (en) 1999-07-30 2000-07-24 System and method for marking of audio data

Applications Claiming Priority (2)

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US36512099A 1999-07-30 1999-07-30
US09/365,120 1999-07-30

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WO2001009889A1 true WO2001009889A1 (fr) 2001-02-08

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1624671A1 (fr) * 2003-05-12 2006-02-08 Seiko Epson Corporation Systeme de commande d'enregistrement

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US4704730A (en) * 1984-03-12 1987-11-03 Allophonix, Inc. Multi-state speech encoder and decoder
EP0722225A2 (fr) * 1994-11-17 1996-07-17 Deutsche Thomson-Brandt Gmbh Codage de signaux audio utilisant des spectres de courte durée et un modèle psycho-acoustique
US5613004A (en) * 1995-06-07 1997-03-18 The Dice Company Steganographic method and device
FR2740897A1 (fr) * 1995-11-06 1997-05-09 Aeta Applic Electroniques Tech Procede et dispositif d'identification de donnees audio et/ou video, sans introduire de perturbations perceptibles
WO2000022811A1 (fr) * 1998-10-14 2000-04-20 Canon Sales Co., Inc. Authentification de documents a l'aide de marques separees de l'information qu'ils contiennent
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WO2000022811A1 (fr) * 1998-10-14 2000-04-20 Canon Sales Co., Inc. Authentification de documents a l'aide de marques separees de l'information qu'ils contiennent
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Cited By (2)

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Publication number Priority date Publication date Assignee Title
EP1624671A1 (fr) * 2003-05-12 2006-02-08 Seiko Epson Corporation Systeme de commande d'enregistrement
EP1624671A4 (fr) * 2003-05-12 2007-05-02 Seiko Epson Corp Systeme de commande d'enregistrement

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