Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
  • H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
  • H04N21/23—Processing of content or additional data; Elementary server operations; Server middleware
  • H04N21/233—Processing of audio elementary streams
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
  • H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
  • H04N21/23—Processing of content or additional data; Elementary server operations; Server middleware
  • H04N21/234—Processing of video elementary streams, e.g. splicing of video streams or manipulating encoded video stream scene graphs
  • H04N21/2347—Processing of video elementary streams, e.g. splicing of video streams or manipulating encoded video stream scene graphs involving video stream encryption
  • H04N21/23476—Processing of video elementary streams, e.g. splicing of video streams or manipulating encoded video stream scene graphs involving video stream encryption by partially encrypting, e.g. encrypting the ending portion of a movie
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
  • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
  • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
  • H04N21/439—Processing of audio elementary streams
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
  • H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client 
  • H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
  • H04N21/631—Multimode Transmission, e.g. transmitting basic layers and enhancement layers of the content over different transmission paths or transmitting with different error corrections, different keys or with different transmission protocols
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
  • H04N21/80—Generation or processing of content or additional data by content creator independently of the distribution process; Content per se
  • H04N21/81—Monomedia components thereof
  • H04N21/8106—Monomedia components thereof involving special audio data, e.g. different tracks for different languages
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
  • H04N7/00—Television systems
  • H04N7/16—Analogue secrecy systems; Analogue subscription systems
  • H04N7/167—Systems rendering the television signal unintelligible and subsequently intelligible
  • H04N7/1675—Providing digital key or authorisation information for generation or regeneration of the scrambling sequence

Definitions

• the present invention relates to the field of processing digital audio streams. It is proposed in the present invention to provide a system for auditory scrambling and recomposing digital audio content.
• the present invention relates more particularly to a device capable of securely transmitting a set of audio streams of high auditory quality to a music or speech player to be recorded in the memory or on the hard drive of a decoder unit connecting the remote transmission network to the audio player, while preserving the auditory quality, but avoiding any fraudulent use such as the possibility of making pirated copies of audio programs recorded in the memory or on the hard disk of the decoder unit.
• the invention relates to a method for distributing digital audio sequences according to a nominal stream format consisting of a succession of frames each comprising at least one digital audio block grouping together a certain number of coefficients corresponding to simple audio elements digitally coded in a mode. specified inside the stream concerned and used by all audio decoders capable of playing it in order to be able to decode it correctly.
• This process includes:
• Said complementary information is defined as a set consisting of data (for example coefficients describing the original digital stream or extracts from the original stream) and functions (for example, the substitution or permutation function).
• a function is defined as containing at least one instruction relating data and operators. Said additional digital information describes the operations to be carried out to recover the original flow from the modified flow.
• the original stream is reconstituted on the recipient equipment from the modified main stream already present or sent in real time to the recipient equipment and additional information, sent in real time at the time of listening, comprising data and functions executed using digital routines (set of instructions).
• a first signal processing circuit for inserting a redundant portion into a portion between adjoining frames and compressing the frames in base time in response to the redundant portions during encoding, a circuit generating a signal for inserting a control signal other than audio information in the redundant portions, a control signal detection circuit for detecting the control signal during decoding and a second signal processing circuit for removing the redundant portions in synchronism with the signal control detected and decompressing the frames in base time in response to the redundant portions.
• Multimedia Adaptive Scrambling System a system for scrambling digital samples representing multimedia data (audio and video), so that the content of these samples is degraded, but recognizable, or otherwise provided with the required quality.
• the quality level is linked to an associated signal / noise ratio, and is determined using objective and subjective tests.
• a given number of Least Significant Bits (LSBs) is scrambled frame by frame, adaptively according to the dynamics of the possible values.
• All encryption keys are included in the audio / video stream and used by the decoder to descramble and restore the stream. After descrambling, the encryption key cannot be recovered, since it is itself scrambled by the decoder.
• the state of the art demonstrates many audio stream protection systems, essentially based on data encryption, by adding encryption keys independent of the content of the audio stream, and which therefore modify the format of the structured stream.
• a particular and different embodiment is that of the company Coding Technologies, which consists in protecting by scrambling a selected part of the bitstream (the bitstream at the output of the audio encoder is called "bitstream") and not the entire bitstream.
• Protected areas represent spectral values of the audio signal, leading to the fact that during decoding without decryption, the audio stream is distorted and unpleasant to listen to.
• the present invention intends to remedy the drawbacks of the prior art by proposing to apply adaptive and progressive scrambling as a function of the structure of the audio stream, the profile of the client and external events.
• the term “scrambling” is understood to mean the modification of a digital audio stream by appropriate methods so that this stream remains in conformity with the standard or standard with which it has been digitally encoded, while making it audible by an audio player (or player), but altered from the point of view of human auditory perception.
• the term “descrambling” is understood to mean the process of restitution by appropriate methods of the initial stream, the audio stream restored after descrambling being identical to the original original audio stream.
• the original flow is reconstituted on the recipient equipment from the modified main flow already present or sent in real time to the recipient equipment and additional information, sent in real time at the time of listening, including data and functions executed using digital routines (set of instructions). All or part of the additional information is sent according to the client's profile and rights.
• the quantity of information contained in said sub-part of the additional information is defined as the number of data and / or functions belonging to the additional information sent to the recipient during the connection.
• the type of information contained in said subpart corresponds to a level of scalability determined in depending on the recipient's profile.
• the type of data relates to the habits of the recipient (time of connection, duration of connection, regularity of connection and payments), his environment (lives in a big city, the weather right now ) and its characteristics (age, sex, religion, community).
• Said additional information is composed of at least functions, which are personalized for each recipient in relation to the connection session.
• a session is defined from the connection time, the duration, the type of said modified stream listened to and the connected elements (recipients, servers).
• Said additional information is subdivided into at least two sub-parts, each of the sub-parts being able to be distributed by different media, or by the same media. For example, in the case of distribution of additional information by several media, we can ensure more complex management of the rights of recipients.
• the “profile” of the user is understood to mean a digital file comprising descriptors and information specific to the user, for example his cultural preferences and his social and cultural characteristics, his usage habits such as the frequency of use.
• audio means, the average duration of listening to a scrambled audio sequence, the frequency of listening to a scrambled sequence, the price that the user is ready to pay or any other behavioral characteristic with regard to exploitation audio clips.
• This profile is formalized by a digital file or a digital table usable by computer means.
• Many scrambling systems have an immediate effect, either the initial flow is fully scrambled, or the initial flow is not at all scrambled, so generally different audio sequences can be scrambled with the same algorithm and setting parameters. Many protections used do not change the scrambling of an audio stream according to its content.
• adaptive and progressive scrambling is applied as a function of the structure of the audio bitstream and / or its content, by changing the algorithms and / or the scrambling parameters as a function of the characteristics of the audio stream. and the user application, and this in order to achieve reliable protection, from the point of view of the deterioration of the original lux and resistance to piracy, for a minimal cost, while ensuring in the end the quality of service required by the recipient or the customer.
• Different adaptations of the scrambling are applied, such as those mentioned below.
• the invention relates in its most general sense to a method for the distribution of digital audio sequences according to a nominal stream format constituted by a succession of frames each comprising at least one digital audio block grouping together a plurality of coefficients corresponding to simple audio elements digitally coded, the method comprising a step of modifying at least one block of the original flow characterized in that said modifying step acts adaptively on said original flow as a function of at least part of the characteristics representative of the structure, the content and parameters of the original audio stream, the recipient's profile, and external events.
• the modification step consists in replacing part of said coefficients to produce on the one hand a main audio stream in nominal format and on the other hand complementary modification information allowing the reconstruction of the original stream by a decoder of the recipient equipment, the scope of the modifications being variable and determined by said representative characteristics.
• the modified main stream is recorded on the recipient equipment before the transmission of the additional information on the recipient equipment.
• the modified main stream is recorded on a physical medium to be transmitted to the recipient equipment prior to the transmission of the additional information on the recipient equipment.
• the modified main stream and the additional information are transmitted together in real time.
• said additional modification information comprises at least one digital routine capable of executing a function.
• said additional modification information is subdivided into at least two sub-parts.
• said sub-parts of the additional modification information are distributed by different media.
• said sub-parts of the additional modification information are distributed by the same medium.
• the additional information is transmitted on a physical vector.
• the additional information is transmitted online.
• said digital audio sequences are modified in a differentiated manner as a function of their audio content.
• said digital audio sequences are modified in a differentiated manner as a function of the modified scalability layer.
• said digital audio sequences are modified in a differentiated manner as a function of the bit rate in kilo bits per second (kbits / s) of the original stream.
• said digital audio sequences are modified in a differentiated manner as a function of the profile and the digital level defined by the standard or standard with which or which they were encoded.
• said digital audio sequences are modified in a differentiated manner as a function of the number of audio channels present in the stream.
• said digital audio sequences are modified in a differentiated manner as a function of the coupling and the multiplexing between the different audio channels present in the stream.
• said digital audio sequences are modified in a differentiated manner as a function of the sampling frequency with which the audio stream has been encoded.
• said digital audio sequences are modified in a differentiated manner according to the psychoacoustic model used.
• said digital audio sequences are modified in a differentiated manner as a function of their granular scalability.
• said digital audio sequences are modified in a progressive manner increasing the degradation effect until the audio stream is completely scrambled.
• said digital audio sequences are modified with a random generation of scrambling parameters and configurations.
• the method comprises a prior step of analog / digital conversion in a structured format, the method being applied to an analog audio signal.
• the present invention also relates to a system for the distribution of digital audio sequences comprising an audio server comprising means for broadcasting a stream modified in accordance with any one of the preceding methods, and a plurality of equipment provided with a circuit 'scrambling, characterized in that the server further comprises a means for recording the digital profile of each recipient and a means for controlling the modification means as a function of input variables corresponding to at least some of the characteristics representative of the structure, content and parameters of the original audio stream, the recipient's profile, and external events.
• a digital audio stream is generally composed of sequences made up of frames or blocks, organized according to a specific digital format for each audio coder, including the headers of the frames with the different encoding parameters and coefficients relative to a specific representation of the digital audio samples. Knowing how the audio signal is modeled, compressed and encoded for the audio coder and / or the given standard or standard, it is always possible to extract from the bitstream the main parameters which describe it and which are sent to the decoder. Once these parameters have been identified, they are modified so that the audio stream generated by the coder and / or the given standard conforms to this coder and / or this standard. In addition, the modification ensures the stability of the sound signal, but makes it unusable by the user, because it is scrambled. However, it can be understood and interpreted in the decoder corresponding to its encoding and played by a player without the latter being disturbed.
• the modification of one or more of the components of said audio signal will cause its degradation from the point of view auditory and transform it into a completely incomprehensible and unpleasant signal from the point of view of subjective auditory perception.
• the part of the audio signal or the component describing it which will be modified depends on its encoding, for each coder-decoder given, and this whether for speech, music, noise or special effects, synthetic sounds or any signal audio of the same type. According to the way in which the encoding and the transmission of the resulting parameters are carried out, one can have direct or indirect information on the main characteristics of the audio signal and therefore modify them. This principle is applicable for all types of digital coders as well as for all their base and enhancement layers or the combination of both.
• An adaptation of the scrambling parameters is applied as a function of the content of the audio stream: natural or synthetic speech, music, noise, natural or synthetic or compound sounds, special effects.
• the HVXC (Harmonie Vector excitation Coding) encoder for speech, and the HILN (Harmonie and Individual Lines plus Noise) encoder for music defined by the MPEG-4 standard are parametric encoders that encode the audio signal separately or jointly depending on its content.
• the bitstream coming from the HVXC contains the values of the LSP (Line Spectral Pairs) reflecting the LPC (Linear Predictive Coding) parameters.
• the values of the LSPs of the current frame are vectorized in two stages, are stabilized in a value in order to ensure the stability of the LPC synthesis filter and are then stored in a bitstream in ascending order, with a minimum of distance between adjacent coefficients .
• the indices of the vectorally quantified LSP pairs are transmitted to the decoder, which restores the values of the LSPs and therefore of the LPCs from standard tables. By replacing the original indices with other values taken from predefined tables in the standard, the bitstream will remain compliant, but the decoded LSP values will not correspond to the original LPC parameters. Consequently, the spectral envelope will be modified and the speech deteriorated.
• the parameters for the base layer, for the improvement layer or for both layers are modified.
• An adaptation is also applied as a function of the bit rate in number of kilo bits per second (kbits / s) of the audio stream, whether constant or variable.
• kbits / s the bit rate in number of kilo bits per second (kbits / s) of the audio stream, whether constant or variable.
• For some more complex audio streams such as those of the MPEG-4 type, which have a variable bit rate in very large proportions (from 2 kbits / s to 64 its / s), we choose the scrambling parameters according to the bit rate , since scrambling for a low bit rate of the order of 2 kbits / s is less effective for higher bit rates, where the encoding accuracy is much higher.
• the AAC (Advanced Audio Coding) encoding scheme with BSAC (Bit Sliced Arithmetic Coding) provides the possibility of noise reduction encoding from an AAC bitstream to a bitstream with fine granular scalability between 16 kbits / s and 64 kbits / s per channel, whose bit rate is adjustable with a step of 1 kbits / s.
• adaptive scrambling is applied according to the types of objects contained in the stream, the profile ("profile"), the level (“level”) , designating the complexity and the options used when building the audio stream. Indeed, in the context of MPEG-4 audio, there are a multitude of audio objects and profiles.
• one of the profiles is the “Simple scalable” which contains the CELP (Code Excited Linear Prediction) and AAC (Advanced Audio Coding) tools. Scrambling is performed according to the parameters of these two encoders. The adaptive modification of the elements of the audio stream is carried out according to the types of audio objects that each profile and level contains. An adaptation of the scrambling parameters is also applied as a function of the number of audio channels present in the stream.
• CELP Code Excited Linear Prediction
• AAC Advanced Audio Coding
• An adaptation of the scrambling parameters is applied as a function of the coupling and the multiplexing between the different audio channels present in the stream.
• An adaptation of the scrambling parameters is applied as a function of the sampling frequency with which the audio stream has been encoded.
• An adaptation of the scrambling parameters is applied as a function of the psychoacoustic model used, characterizing certain audio encoders.
• the psycho-acoustic model estimates the thresholds determining the maximum quantization error which can be admitted during compression while preserving the audio quality.
• the spectral data are quantified and coded according to these estimated thresholds.
• the quantification is chosen according to the estimated thresholds, for example the quantification can be uniform or non-uniform, and it is carried out using scale factors. By modifying the values of these scale factors, coded in differential in the bit stream, one introduces a quantification error, because the scale factors no longer correspond to those defined by the estimates of the psychoacoustic model.
• the first scale factor when we want to obtain a strong hearing impairment, we modify the first scale factor, since all the scale factors are coded in differential with respect to the first scale factor, all the values which follow are erroneous and the audio signal is highly disturbed. Progressive scrambling is also applied, so that the user begins to listen to the unscrambled audio stream. Then, we start with a light scrambling which we reinforce more and more, until the audio stream becomes completely scrambled. The aim is to arouse the user's interest in the audio stream, but by taking away their rights to listen to it if they have not purchased it.
• One embodiment of this application is to scramble the audio stream with one or more of the algorithms given by gradually modifying the scrambling parameters over a determined period of time so as to increase the inconvenience, until arriving at a completely scrambled stream and inaudible.
• Adaptive scrambling is generally carried out as a function of the content, characteristics, structure and composition of the digital stream defined by a given standard or standard.
• Scrambling is also performed with a random generation of the parametric combinations to apply for scrambling the audio stream.
• a robust protection is provided which is difficult to attack or which cannot be hacked by a malicious person.
• An adaptation of the scrambling parameters and algorithms is also applied as a function of external events, such as the time of broadcast, the audience rate, social-political events, or disturbances during the broadcast.
• FIG. 1 illustrates a particular embodiment of the client-server system according to the invention.
• the MPEG-AAC type audio stream that one wishes to secure (1) is sent to an analysis (121) and scrambling (122) system which will generate a modified main stream and additional information at the output.
• the original stream (1) can be directly in digital form (10) or in analog form (11). In the latter case, the analog stream (11) is converted by an encoder not shown in a digital format (10). In the following the text, we will note (1) the digital audio input stream.
• the complementary information (123), in any format, contains the references of the parts of the audio samples which have been modified and is placed in the buffer (126).
• the analysis (121) and scrambling (122) system decides which adaptive scrambling to apply and which parameters of the flow to modify and also according to the rights of the client, from which way to apply the modifications, for example progressive or not.
• the MPEG-AAC stream (125) is then transmitted, either in physical form on a CD-ROM, non-volatile memory, DVD, etc., or via a network (4) of the telephone network type, DSL (Digital Subscriber Line) , BLR (Local Radio Loop), DAB (Digital Audio Broadcasting), RTC (Switched Telephone Network), digital mobiles (GSM, GPRS, UMTS), wireless, cable, satellite, etc., to the customer (8), and more specifically in its memory (81) of RAM, ROM, hard disk type.
• DSL Digital Subscriber Line
• BLR Land Radio Loop
• DAB Digital Audio Broadcasting
• RTC Switchched Telephone Network
• GSM Global System for Mobile Communications
• GPRS Portable Network
• UMTS Universal Mobile Communications Service
• the recipient (8) does not have the necessary rights to play the audio sequence.
• the stream (125) generated by the scrambling system (122) present in its memory (81) is passed to the synthesis system (82), which does not modify it and transmits it identically to a reader.
• classical audio (83) and its content, strongly hearing impaired, is played by the player (83) on the headphones or speakers (9). or, the recipient (8) has the rights to listen to the audio sequence.
• the server (12) transmits the appropriate additional information (126) via the link (6), corresponding to the type of scrambling carried out.
• the synthesis system makes a request for audition to the server (12) containing the information necessary (126) for the recovery of the original audio sequence (1).
• the server (12) then sends via the link (6) via transmission networks such as analog or digital telephone lines, DSL (Digital Subscriber Line), BLR (Local Radio Loop), DAB (Digital Audio Broadcasting), PSTN (Telephone Network Switched), digital mobile networks (GSM, GPRS, UMTS), wireless, cable or satellite additional information (126) allowing the reconstruction of the audio sequence so that the client (8) can listen and / or store the audio sequence.
• the synthesis system (82) then proceeds to descramble the audio sequence by reconstructing the original stream by combining the modified main stream (125) and the additional information (126).
• the audio stream thus obtained at the output of the synthesis system (82) is then transmitted to the conventional audio player (83) and the original audio sequence is played on the headphones or the speakers (9).
• More and more coders have the option of operating at variable bit rates in order to satisfy specific applications, such as for example to respond limited bandwidth constraints.
• An example of an encoder intended to ensure acceptable quality for speech, while respecting low bandwidth is the AMR ("Adaptive Multi Rate" in English) encoder, designed for cell phones, which can operate in eight different modes, the speed of which varies between 4.75 kbits / s and 12.2 kbits / s.
• the present invention makes differentiated modifications according to the mode with which the audio stream was encoded, that is to say according to the bit rate, the length of the respective components of the frame, as well as according to the degree of hearing impairment desired.
• the structure of the AMR frame is as follows: - The indexes corresponding to the frequency spectral pairs, called LSF ("Line Spectral Frequencies" in English), relating to the LSP parameters ("Line Spectral Pairs "in English), therefore also with LPC parameters (" Linear Predictive Coding "in English), that is to say with the form of the formants filter, said indexes being common to the entire frame;
• modifying the value of the fundamental delay by substitution with a different value causes a frequency shift: a lower value causes a distortion of the voice, the effect obtained is a muffled sound, with crackling sounds similar to an “extinction of the voice " .
• the differentiated modifications of the LSF give little additional information, for significant hearing impairment. Preferably, they are combined with other modifications.
• the signs of the pulsations relating to the construction of the excitation are modified.
• the excitation is also changed and the sound is totally distorted.
• the frame structure is similar with the difference that it contains a single set of three LSFs; differentiated modifications are then applied, taking account of this particularity and of the frame length corresponding to this mode.
• the structure of the frame is slightly different, it does not contain the amplitude of the fundamental, nor the gain of the tables with fixed values, but a set of gains relating to the tables of fixed and adaptive values, used to scale the excitement built from the addition of adaptive and innovation vector codes.
• the modifications applied take these specific features into account. Modifying the LSF produces significant degradation, however, since the audio bit rates are low, small modifications are enough to obtain a strong hearing degradation.
• the differentiated modifications are made taking into account the desired bit rate for the additional information.
• the present invention is not limited to the modifications cited as exemplary embodiments, said modifications guaranteeing that the amplitude values of the sound authorized are not exceeded and guaranteeing the conformity of the modified main stream with the original audio stream.
• the reconstituted stream is auditory identical to the original, but different from the binary point of view of the original stream, and this in order to strengthen security.
• the reconstituted flow is strictly identical to the original, the process is lossless.

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• Engineering & Computer Science (AREA)
• Multimedia (AREA)
• Signal Processing (AREA)
• Computer Security & Cryptography (AREA)
• Compression, Expansion, Code Conversion, And Decoders (AREA)
• Signal Processing For Digital Recording And Reproducing (AREA)

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• 2002
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