EP1587219A2 - Record carrier having an encoded wideband digital audio signal recorded on it - Google Patents

Abstract

Reproduction accuracy of a digital signal, representing for example stereoaudio signals, is improved by transmitting sample data, which have been encoded to formtransmission signals. The stereo audio signals are coded into sub(band) signals.Corresponding sub(band)signals of the stereo audio signals may be processed to obtaincomposite sub(band) signals, which are recorded on a record carrier.

To identify whether corresponding sub(band) signals have been processed ornot, a mode indicator control signal indicating whether a corresponding sub signal of the atleast two signal components have or have not been processed into a composite sub signal isalso being recorded on the record carrier.

Description

BACKGROUND OF THE INVENTIONFIELD OF THE INVENTION

The invention relates to a record carrier having an encoded wide-band digitalaudio signal recorded on it, the wide-band digital audio signal comprising at least a first and asecond signal component, the signal components being filtered into sub signals for the atleast two signal components, a sub signal comprising sample information.

DESCRIPTION OF THE RELATED ART

A record carrier as defined in the opening paragraph can be used intransmission systems such as they are e.g. known from the article "The Critical Band Coder -Digital Encoding of Speech signals based on the Percentual requirements of the AuditorySystem" by M.E. Krasner in Proc. IEEE ICASSP 80, Vol. 1, pp.327-331, April 9-11, 1980.This article relates to a transmission system in which the transmitter (recording device)employs a subband coding system and the receiver (reproducing device) employs acorresponding subband decoding system, but the invention is not limited to such a codingsystem, as will become apparent hereinafter.

In the system known from this publication, the speech signal band is dividedinto a plurality of sub-bands having bandwidths approximately corresponding to thebandwidths of the critical bands of the human ear in the respective frequency ranges(cf. Fig. 2 in the article of Krasner). This division has been selected because, on the ground ofpsycho-acoustic experiments, it is foreseeable that the quantization noise in such a sub-bandwill 10 be masked to an optimum extent by the signals in this sub-band if, in the quantization,allowance is made for the noise-masking curve of the human ear (this curve giving thethreshold value for noise masking in a critical band by a single tone in the center of thecritical band, cf. Fig. 3 in the article by Krasner).

It should, however, be noted that the invention is not restricted to an encodinginto sub-band signals. It is equally well possible to apply transform coding in the encoder, atransform coding being described in the publication "Low bit-rate coding of high-quality audio signals. An introduction to the MASCAM system" by G. Theile et al., in EBUTechnical Review, No. 230 (August 1988).

In the case of a high-quality digital music signal, which, in conformity withthe Compact Disc Standard, is represented by 16 bits per signal sample in the case of asample frequency of 1/T = 44.1 kHz, it is found that with a suitably selected bandwidth and asuitably selected quantization for the respective sub-bands, the use of this known sub-bandcoding system yields quantized output signals of the coder which can be represented by anaverage number of approximately 2.5 bits per signal sample, the quality of the replica of themusic signal not differing perceptibly from that of the original music signal in substantiallyall passages of substantially all kinds of music signals.

The sub-bands need not necessarily correspond to the bandwidths of thecritical bands of the human ear. Alternatively, the sub-bands may have other bandwidths, forexample, they may all have the same bandwidth, provided that allowance is made for this indetermining the masking threshold.

The known record carrier has the disadvantage that, in some cases perceptibledifferences occur in the signal reproduced, such perceptible differences being in the form of adistortion component present in the signal reproduced from the record carrier.

SUMMARY OF THE INVENTION

It is an object of the invention to provide measures to enable the transmissionof the wideband digital signal via a record carrier so as to realize a significant reduction ofthe distortion component present in the signal reproduced from the record carrier.

This object is achieved by the record carrier as claimed in the appendedclaims.

The invention is based on the recognition that the distortion arises because ofthe fact that sometimes the specific number of bits being available for the quantization of thewideband digital signal in the recording device is too low. As a result, a number of bits isallocated to a sub(band) signal, which is too low. This results in a quantization of a sub-signal,which is too rough, leading to audible distortion upon reproduction and afterdecoding. By processing corresponding sub-signals (subband signals) into a compositesignal, it suffices to quantize only one composite sub(band) signal instead of the twocorresponding sub(band) signals. This results in less bits being required in the quantizationstep for the quantization of the sub(band) signal. As an alternative, it offers the composite signal to be quantized with a relatively largernumber of bits than if the two sub(band) signals would have been quantized separately.

It is a further object of the invention to provide a number of steps for thetransmission system, in particular, a very specific choice for the format with which the digitalwide-band signal, after conversion into the second digital signal, can be transmitted via thetransmission medium, in such a way that a flexible and highly versatile transmission systemis obtained. This is to be understood to mean that the transmitter should be capable ofconverting wide-band digital signals of different formats (these formats differing, inter alia,with respect to the sample frequency F_S of the wide-band digital signal, which may havedifferent values, such as, 32 kHz, 44. 1 kHz and 48 kHz, as laid down in the digital audiointerface standard of the AES and the EBU) into thesecond digital signal. Similarly, the receiver should be capable of deriving a wide-band signalof the correct format from said second digital signal. To this end, the transmission system inaccordance with the invention is characterized in that if P in the formulaP = BR x nS / N x FSis an integer, where BR is the bit rate of the second digital signal, and n_S is the number ofsamples of the wideband digital signal whose corresponding information, which belongs tothe second digital signal, is included in one frame of the second digital signal, the number ofinformation packets B in one frame is P, and in that, if P is not an integer, the number ofinformation packets in a number of the frames is P', P' being the next lower integer followingP, and the number of information packets in the other frames is equal toP'+1 so as to exactlycomply with the requirement that the average frame rate of the second digital signal shouldbe substantially equal to F_S/n_S, and that a frame should comprise at least a first frame portionincluding the synchronizing information. The purpose of dividing the frames into Binformation packets is that, for a wide-band digital signal of an arbitrary sample frequencyF_S, the average frame rate of the second digital signal transmitted by the transmitter is nowsuch that the duration of a frame in the second digital signal corresponds to the durationoccupied by n_S samples of the wide-band signal. Moreover, this enables the synchronizationto be maintained on an information-packet basis, which is simpler and more reliable thanmaintaining the synchronization on a bit basis. Thus, in those cases where P is not an integer,the transmitter is capable, at instants at which this possible and also necessary, to provide aframe withP'+1 instead of P' information blocks, so that the average frame rate of the second digital signal can be maintained equal to F_S/n_S. Since, in this case, the spacing between thesynchronizing information (synchronizing signals or synchronizing words) included in thefirst frame portion of succeeding frames is also an integral multiple of the length of aninformation packet, it remains possible to maintain the synchronization on an informationpacket basis. Preferably, the first frame portion further contains information related to thenumber of information packets in a frame. In a frame comprising B information packets, thisinformation may be equal to the value B. This means that this information corresponds to P'for frames comprising P' information packets, and toP'+1 forframes comprising P'+1information packets. Another possibility is that this information corresponds to P' for allframes, regardless of whether a frame comprises P' orP'+1 information packets. Theadditionally inserted (P'+1)th information packet may comprise, for example, merely "zeros".In that case, this information packet does not contain any useful information. Of course, theadditional information packet may also be filled with useful information. The first frameportion may further comprise system information. This may include the sample frequency F_Sof the wide-band digital signal applied to the transmitter, copy-protection codes, the type ofwide-band digital signal applied to the transmitter, such as a stereo-audio signal or a mono-audiosignal, or a digital signal comprising two substantially independent audio signals.However, other system information is also possible, as will become apparent hereinafter.Including the system information makes it possible for the receiver to be also flexible andenables the received second digital signal to be correctly reconverted into the wide-banddigital signal. The second and the third frame portions of a frame captain signal information.The transmitter may comprise a coder comprising signal-splitting means responsive to thewide-band digital signal to generate a second digital signal in the form of a number of M sub-signals,M being larger than 1, and comprising means for quantizing the respective sub-signals.For this purpose, an arbitrary transform coding, such as the fast Fourier transform(FFT), may be used. In that case, the transmission system is characterized in that the secondframe portion of a frame contains allocation information which, for at least a number of sub-signals,indicates the number of bits representing the samples of the quantized sub-signalsderived from said sub-signals, and in that the third frame portion contains the samples of atleast said quantized sub-signals (if present). At the receiving end, it is then necessary to applyan inverse transform coding, for example, an inverse Fourier transform (1FFT), to recover thewide-band digital signal. The transmission system, in which the signal-splitting means takesthe form of analysis-filter means responsive to the wide-band digital signal to generate anumber of M sub-band signals, this analysis-filter means dividing the signal band of the wide-band digital signal, using a sample-frequency reduction, into successive sub-bandshaving band numbers m increasing with the frequency, and in which the quantization meansis adapted to quantize the respective sub-band signals block by block, is a system employingsub-band coding as described above. Such a transmission system is characterized further inthat, for at least a number of the sub-band signals, the allocation information in the secondframe portion of a frame specifies the number of bits representing the samples of thequantized sub-band signals derived from said sub-band signals, and in that the third frameportion contains the samples of at least said quantized sub-band signals (if present). Thismeans, in fact, that the allocation information is inserted in a frame before the samples. Thisallocation information is needed to enable the continuous serial bit stream of the samples inthe third frame portion to be subdivided into the various individual samples of the correctnumber of bits at the receiving end. The allocation information may require that all samplesare represented by a fixed number of bits per sub-band per frame. This is referred to as atransmitter based on fixed or static bit allocation. The allocation information may also implythat a number of bits variable in time is used for the samples in a sub-band. This is referred toas a transmitter based on the system of adaptive or dynamic bit allocation. Fixed and adaptivebit allocations are described, inter alia, in the publication "Low bit-rate coding of high qualityaudio signals. An introduction to the MASCAM system" by G. Theile et al., EBU TechnicalReview, No. 230 (August 1988). Inserting the allocation information in a frame before thesamples in a frame, has the advantage that, at the receiving end, a simpler decoding becomespossible, which can be carried out in real time and which produces only a slight signal delay.As a result of this sequence, it is no longer necessary to first store all the information in thethird frame portion in a memory in the receiver. Upon arrival of the second digital signal, theallocation information is stored in a memory in the receiver. Information content of theallocation information is much smaller than the information content of the samples in thethird frame portion, so that a substantially smaller store capacity is needed than in the casethat all the samples would have to be stored in the receiver. Immediately upon arrival of theserial data stream of the samples in the third frame portion, this data stream can be dividedinto the various samples having the number of bits specified by the allocation information, sothat no previous storage of the signal information is necessary. The allocation information forall the sub-bands can be included in a frame. However, this is not necessary, as will becomeapparent hereinafter.

The transmission system may be characterized further in that, in addition, thethird frame portion includes information related to scale factors, a scale factor being associated with at least one of the quantized sub-band signals contained in the third frameportion, and in that the scale factor information is included in the third frame portion beforethe quantized sub-band signals. The samples can be coded in the transmitter without beingnormalized, i.e., without the amplitudes of a block of samples in a sub-band having beendivided by the amplitude of the sample having the largest amplitude in this block. In thatcase, no scale factors have to be transmitted. If the samples are normalized during coding,scale factor information has to be transmitted to provide a measure of said largest amplitude.If, in this case, the scale factor information is also inserted in the third frame portion beforethe samples, it is possible that during reception, the scale factors to be derived from said scaleinformation are first stored in a memory and the samples are multiplied immediately uponarrival, i.e., without a time delay, by the inverse values of said scale factors. The scale factorinformation may be constituted by the scale factors themselves. It is obvious that a scalefactor as inserted in the third frame portion may also be the inverse of the amplitude of thelargest sample in a block, so that in the receiver, it is not necessary to determine the inversevalue and, consequently, decoding can be faster. Alternatively, the values of the scale factorsmay be encoded prior to insertion in the third frame portion as scale factor information andsubsequent transmission. Moreover, it is evident that if, after quantization in the transmitter,the sub-band signal in a sub-band is zero, which obviously will be apparent from theallocation information for the sub-band, no scale factor information for this sub-band has tobe transmitted. The transmission system, in which the receiver comprises a decodercomprising synthesis filter means responsive to the respective quantized sub-band signals toconstruct a replica of the wide-band digital signal, this synthesis filter means combining thesub-bands applying sample-frequency increase to form the signal band of the wide-banddigital signal, may be characterized in that the samples of the sub-band signals (if present) areinserted in the third frame portion in a sequence corresponding to the sequence in which saidsamples are applied to the synthesis filter means upon reception in the receiver. Inserting thesamples in the third frame portion in the same sequence as that in which they are applied tothe synthesis filter means in the receiver also results in fast decoding, which again does notrequire additional storage of the samples in the receiver before they can be further processed.Consequently, the storage capacity required in the receiver can be limited substantially to thestorage capacity needed for the storage of the system information, the allocation informationand, if applicable, the scale factor information. Moreover, a limited signal delay is produced,which is mainly the result of the signal processing performed upon the samples. Theallocation information for the various quantized sub-band signals is suitably inserted in the second frame portion in the same sequence as that in which the samples of the sub-bandsignals are included in the third frame portion. The same applies to the sequence of the scalefactors. If desired, the frames may also be divided into four portions, the first, the second andthe third frame portions being as described hereinbefore. The last (fourth) frame portion inthe frame may then contain error-detection and/or error-correction information. Uponreception of this information in the receiver, it is possible to apply a correction for errorsproduced in the second digital signal during transmission. As already stated, the wide-banddigital signal may be a monophonic signal. Alternatively, the wide-band digital signal may bea stereo audio signal made up of a first (left) channel component and a second (right) channelcomponent. If the transmission system is based on a sub-band coding system, the transmitterwill supply sub-band signals each comprising a first and a second sub-band signalcomponent, which, after quantization in the quantization means, are converted to form firstand second quantized sub-band signal components. In this case, the frames should alsoinclude allocation information and scale-factor information (if the samples have been scaledin the transmitter). The sequence is also important here. It is obvious that the system can beextended to handle a wide-band digital signal comprising more than two signal components.

The inventive steps may be applied to digital transmission systems, forexample, systems for the transmission of digital audio signals (digital audio broadcast) viathe ether. However, other uses are also conceivable. An example of this is a transmission viaoptical or magnetic media. Optical-media transmissions may be, for example, transmissionsvia glass fibers or by means of optical discs or tapes. Magnetic-media transmissions arepossible, for example, by means of a magnetic disc or a magnetic tape. The second digitalsignal is then stored in the format as proposed by the invention in one or more tracks of arecord carrier, such as an optical or magnetic disc or a magnetic tape. The versatility andflexibility of the transmission system thus resides in the special format with which theinformation in the form of the second digital signal is transmitted, for example, via a recordcarrier. This is combined with the special construction of the transmitter, which is capable ofgenerating this special format for various types of input signals. The transmitter generates thesystem information required for every type of signal and inserts this information in the datastream to be transmitted. At the receiving end, this is achieved by means of a specificreceiver, which extracts said system information from the data stream and employs it for acorrect decoding. The information packets then constitute a kind of fictitious units, which areused to define the length of a frame. This means that they need not be explicitly discernible inthe information stream of the second digital signal. Moreover, the relationship of the information packets with the existing digital audio interface standard is as defined in the IECStandard No. 958. This standard, as normally applied to consumer products, defines framescontaining one sample of both the left-hand and the right-hand channels of a stereo signal.These samples are represented by means of 16-bit two's complement words. If N = 32 isselected, one frame of this digital audio interface standard can transmit exactly oneinformation packet of the second digital signal. In the digital audio interface standard, theframe rate is equal to the sample rate. For the present purpose, the frame rate should beselected to be equal to BR/N. This enables the present IC's employed in standard digital audiointerface equipment to be used.

BRIEF DESCRIPTION OF THE DRAWINGS

With the above and additional objects and advantages in mind as willhereinafter appear, the invention will be described with reference to the accompanyingdrawings, in which:

Figs. 1a-1c show a diagram of a digital signal according to the invention,generated by an encoder and made up of frames each composed of information packets;

Fig. 2 is a diagram of the structure of a frame according to a preferredembodiment including scale factors;

Fig. 3 is a diagram of the structure of the first portion of the frame of Fig. 2;

Fig. 4 is a block diagram of a digital transmissionsystem for producing and using a signal according to the invention, comprising a transmitterhaving an encoder and a receiver having a decoder;

Fig. 5 is a table showing the number of informationpackets B in a frame, for certain values of bit rate BR and sample frequency F_S;

Fig. 6 is a table showing the numbers of frames in a padding sequence, and thenumber of frames in that sequence having an additional information packet (a dummy slot)for different bit rates;

Fig. 7 is a table showing the system information included in the first portion ofa frame;

Fig. 8 is a table showing a distribution of information between two channelsfor different modes;

Fig. 9 is a table of meanings of allocation information inserted in the secondportion of a frame;

Figs. 10 and 11 are tables showing sequences in which allocation informationis stored for two different formats;

Fig. 12 is a block diagram of a receiver including a decoder for decodingsignals according to the invention;

Fig. 13 is a simplified block diagram of an encoder for recording a signal on amagnetic record carrier according to the invention;

Fig. 14 is a simplified block diagram of a receiver for producing a replicasignal corresponding to a transmission signal in a magnetic record carrier according to theinvention;

Figs. 15a-15d are diagrams of different arrangements of scale factors andsamples in the third portion of a frame of a transmission signal;

Fig. 16 is a block diagram of a sub-band coding transmitter arrangement;

Fig. 17 is a diagram of another structure for the first portion of a frame;

Fig. 18 is a table showing system information included in the structure of Fig.17;

Fig. 19 is a diagram of a structure for a portion of the structure of Fig. 17,where the signal is an audio signal;

Fig. 20 is a table showing bit codings in an embodiment of the structure of Fig.19 for stereo signals;

Fig. 21 is a table showing a sequence for allocation informationaccommodated in a second frame portion associated with the first portion of Fig. 17;

Figs. 22a-22d are tables showing sequences for allocation informationaccommodated in a second frame portion associated with the first portion of Fig. 17, for astereo intensity mode;

Fig. 23 is a diagram of a frame structure including an additional signal;

Fig. 24 is a binary number diagram relating the sample with largest absolutevalue to an intermediate value used for scale factor computations;

Fig. 25 is a table showing quantization of scaled samples to form q-bit digitalrepresentations; and

Fig. 26 is a table showing dequantization of the q-bit digital representations.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

Fig. 1 shows, diagrammatically, the transmission signal as generated by thetransmitter and transmitted through a transmission medium (real time or via recording). The transmission signal is in the form of a serial digital data stream. The transmission signalcomprises frames, two such frames, i.e., the frame j and the frame j+1, being shown in Fig.1a. The frames, such as the frame j, comprise a plurality of information packets IP1, IP2, IP3,..., such as shown in Fig. 1b. Each information packet, such as IP3, is composed of N bits b_o,b₁, b₂, ..., b_N-1, such as shown in Fig. 1c.

NUMBER OF PACKETS

The number of information packets in a frame depends upon:

(a) the bit rate BR with which the transmission signal is transmitted throughthe transmission medium,

(b) the number of bits N in an information packet, N being larger than 1,

(c) the sample frequency F_S of the wide-band digital signal, and

(d) the number of samples n_S of the wide-band digital signal.

The information which corresponds to these packets, and which afterconversion in the transmitter is in the transmission signal, is included in one frame in thefollowing manner.

The parameter P is computed in conformity with the formula:P = (BR x nS) / (N x FS).If this computation yields an integer for P, the number of information packets B in a framewill be equal to P. If the computation does not result in an integer, some frames willcomprise P' information packets and the other frames will compriseP'+1 information packets.P' is the next lower integer following P. The number of frames comprising P' andP'+1information packets is selected in such a way that the average frame rate is equal to F_S/n_S.

Hereinafter, it is assumed that N=32 and n_s=384. The table in Fig. 5 gives thenumber of information packets (slots) in one frame for these values for N and n_S, and for fourvalues of the bit rate BR and three values for the sample frequency F_S. It is evident that for asample frequency F_S equal to 44.1 kHz, the parameter P is not an integer in all cases and that,consequently, a number of frames comprise 34 information packets and the other framescomprise 35 information packets (when BR is 128 kbit/s). This is also illustrated in Fig. 2.

Fig. 2 shows one frame. The frame is composed of P' information packets IP1,IP2, ..., IP P'. Sometimes the frame is composed ofP'+1 information packets. This is achievedby assigning an additional information packet (dummy slot) to the frames of P'information packets. The second column of the table of Fig. 6 gives the number of frames inthe padding sequence for a sample frequency of 44.1 kHz and the aforementioned four bit rates. The third column specifies those frames of that number of frames in the sequencewhich compriseP'+1 information packets. By subtracting the numbers in the second and thethird columns from each other, this yields the number of frames in the sequence comprisingP' information packets. The (P'+1)th information packet need not contain any information,and may then be composed, for example, only of zeroes.

It is obvious that the bit rate BR is not necessarily limited to the four values asgiven in the tables of Figs. 5 and 6. Other (for example, intermediate) values are alsopossible.

Fig. 2 shows that a frame comprises three frame portions FD1, FD2 and FD3in this order. The first frame portion FD1 contains synchronizing information and systeminformation. The second frame portion FD2 contains allocation information. The third frameportion FD3 contains samples and, when applicable, scale factors of the transmission signal.For a further explanation, it is necessary to first describe the operation of the encoderincluded in a transmitter in a transmission system.

TRANSMISSION SYSTEM

Fig. 4 shows, diagrammatically, a transmission system comprising atransmitter 1 having aninput terminal 2 for receiving a wide-band digital signal S_BB, whichmay be a digital audio signal, for example. In the case of an audio signal, this may be a monosignal or a stereo signal, in which case the digital signal comprises a first (left channel) and asecond (right channel) signal component. In this embodiment, the transmitter comprises acoder for sub-band coding of the wide-band digital signal and the receiver consequentlycomprises a sub-band decoder for recovering the wide-band digital signal.

The transmitter comprises ananalysis filter 3 responsive to the digital wide-bandsignal S_BB to divide the wide-band signal into a plurality M of successive frequencysub-bands having band numbers m, where 1 ≤ m ≤ M, which increase with frequency. All ofthese sub-bands may have the same bandwidth but, alternatively, the sub-bands may havedifferent bandwidths. In that case, the sub-bands may correspond, for example, to thebandwidths of the critical bands of the human ear. Theanalysis filter 3 generatessub-band signals S_SB1 to S_SBM, for the respective sub-bands. The transmitter further comprisescircuits for sample-frequency reduction and block-by-block quantization of the respectivesub-band signals, shown as theblock 9 in Fig. 4.

Such a sub-band coder is known and is described, for example, in theaforementioned publications by Krasner and by Theile et al. For a further description of the operation of the sub-band coder, reference is made to these publications, and also to thepublished European Patent Application EPA 289,080, corresponding to U.S. Patent4,896,362, which are therefore incorporated herein by reference. Such a sub-band coderenables a significant data reduction to be achieved in the signal, which is transmitted to thereceiver 5 through thetransmission medium 4, for example, a reduction from 16 bits persample for the wide-band digital signal S_BB to 4 bits per sample if n_S is 384. This means thatthere are blocks of 384 samples of the wide-band digital signal, each sample having a lengthof 16 bits. If a value M = 32 is assumed, the wide-band digital signal is split into 32 sub-bandsignals in theanalysis filter 3. Now, 32 (blocks of) sub-band signals appear on the 32 outputsof theanalysis filter 3, each block comprising 12 samples (the sub-bands have equal width)and each sample having a length of 16 bits. This means that at the outputs of thefilter 3, theinformation content is still equal to the information content of the block of 384 samples of thesignal S_BB at theinput 2.

Thedata reduction circuit 9 operates on the output of thefilter 3 using theknowledge about masking. At least some of the samples in the 32 blocks of 12 samples, eachblock for one sub-band, are quantized more roughly and can thus be represented by a smallernumber of bits. In the case of static bit allocation, all the samples per sub-band per frame areexpressed in a fixed number of bits. This number can be different for two or more sub-bandsbut it can also be equal for the sub-bands, for example, equal to 4 bits. In the case of dynamicbit allocation, the number of bits selected for every sub-band may differ viewed in time, sothat sometimes even a larger data reduction can be achieved, or a higher quality can beachieved with the same bit rate.

The sub-band signals quantized in theblock 9 are applied to agenerator unit 6.Starting from the quantized sub-band signals, thisunit 6 generates the transmission signal asillustrated in Figs. 1 and 2. This transmission signal, as stated hereinbefore, can betransmitted directly through themedium 4. Preferably, however, this transmission signal isfirst adapted in a signal converter (not shown), such as an 8-to-10 converter. Such an 8-to-10converter is described in, for example, European Patent Application EPA 150,082,corresponding to U.S. Patent 4,620,311. This converter converts 8-bit data words into 10-bitdata words, and enables an interleaving process to be applied. The purpose behind theseprocesses is to enable error correction to be performed on the information received at thereceiving side. De-interleaving, error correction and 10-to-8 conversion are then performed inthe receiver.

FRAME FORMAT

The composition and content of the frames will now be explained in moredetail. The first frame portion FD1 in Fig. 2 is shown in greater detail in Fig. 3. Fig. 3 showsthat the first frame portion consists of exactly 32 bits and is, therefore, exactly equal to oneinformation packet, namely, the first information packet IP1 of the frame. The first 16 bits ofthe information packet form the synchronizing signal (or synchronizing word), and maycomprise, for example, only "ones". Thebits 16 to 31 are system information. Thebits 16 to23 represent the number of information packets in a frame. This number consequentlycorresponds to P', both for the frame comprising P' information packets and for framescomprising the additional informationpacket IP P'+1. P' can be, at most, 254 (1111 1110 inbit notation) in order to avoid resemblance to the synchronizing signal. Thebits 24 to 31provide frame format information.

Fig. 7 gives an example of the arrangement and significance of this frameformat information.Bit 24 indicates the type of frame. In the case of format A, the secondframe portion has another length (a different number of information packets) than in the caseof format B. As will become apparent hereinafter, the second frame portion FD2 in the Aformat comprises 8 information packets, namely, the information packets IP2 to IP9inclusive; and in the B format, it comprises 4 information packets, namely, the informationpackets IP2 to IP5 inclusive. Thebits 25 and 26 indicate whether copying of the informationis allowed. Thebits 27 to 31 indicate the function mode. This means:

a) the channel mode, which indicates the type of wide- band signal (as statedhereinbefore this may be a stereo audio signal, a mono audio signal, or an audio signalcomprising two different signal components for example, representing the same text but intwo different languages). Fig. 8 shows how the signal components are divided between thetwo channels (channel I and channel II) in different channel modes.

b) the sample frequency F_S of the wide-band signal.

c) the emphasis, which may be applied to the wide-band digital signal in thetransmitter. Thevalues 50 and 15 µs are the time constants of the emphasis and CCITT J. Thevalue 17 indicates a specific emphasis standard as defined by the CCITT (ComitéConsultative Internationale de Télégraphie et Téléphonie).

The content of the frame portion FD2 in Fig. 2 will bedescribed in more detail with reference to Figs. 9, 10 and 11. In the A format, the secondframe portion contains eight information packets. This is based on the assumptions that thewide-band digital signal S_BB is converted into 32 sub-band signals (for every signal portion of the digital signal S_BB), and that an allocation word having a length of four bits is assigned toevery sub-band. This yields a total of 64 allocation words having a length of 4bits each, which can be accommodated exactly in eight information packets. In the B format,the second frame portion accommodates the allocation information for only half the numberof sub-bands, so that now the second frame portion comprises only 4 information packets.

Fig. 9 lists a set of meanings of the four-bit allocation words AW. Anallocation word associated with a specific sub-band specifies the number of bits by which thesamples of the sub-band signal in the relevant sub-band are represented after quantization intheunit 9. For example, the allocation word AW that is 0100 indicates that the samples arerepresented by 5-bit words. Moreover, it follows from Fig. 9 that theallocation word 0000indicates that no samples have been generated in the relevant sub- band. This may happen,for example, if the sub-band signal in an adjacent sub-band has such a large amplitude thatthis signal fully masks the sub-band signal in the relevant sub-band. Theallocation word1111 is not used because it closely resembles the sync word in the first information packetIP1.

Fig. 10 indicates the sequence, in the case that the frame format is A, in whichthe allocation words AW j,m associated with the two channels j, where j=I or II, and the 32sub-bands of the sequence number m, m ranging from 1 to 32, are arranged in the secondframe portion. The allocation word AW I,1, belonging to the first sub-band signal componentof the first and lowest sub-band (channel I, sub-band 1), is inserted first. After this, theallocation word AW II,1, belonging to the second sub-band signal component of the first andlowest sub-band (channel II, sub-band 1), is inserted in the second frame portion FD2.Subsequently, the allocation word AW I,2, belonging to the first sub-band signal componentof the second and lowest but one sub-band (channel I, sub-band 2), is inserted in the frameportion FD2. This is followed by the allocation word AW II,2, belonging to the second sub-bandsignal component of the second sub-band (channel II, sub-band 2). This sequencecontinues until the allocation word AW II,4, belonging to the second sub-band signalcomponent of the fourth sub-band (channel II, sub-band 4), is inserted in the second frameportion FD2. The second information packet IP2 (slot 2) of the frame, which is the firstinformation packet in the frame portion FD2 of the frame, is then filled exactly.Subsequently, the information packet IP3 (slot 3) is filled with AW I,5; AW II,5; ... AW II,8.This continues in the sequence as illustrated in Fig. 10, which merely gives the indices j-m ofthe inserted allocation word AW j, m.

Fig. 11 indicates the sequence for the allocation words in the case of a B-formatframe. In this case, only allocation words of thesub-bands 1 to 16 are inserted. Thesequence, similar to that illustrated in Fig. 10, corresponds to the sequence in which theseparate samples belonging to a channel j and a sub-band m are applied to a synthesis filterupon reception in the receiver. This will be explained in greater detail hereinafter.

The serial data stream contains, for example, only frames in conformity withthe A format. In the receiver, the allocation information in each frame is then employed forcorrectly deriving the samples from the information in the third frame portion of said frame.The serial data stream may also comprise, more or less alternately, both frames in conformitywith the A format and frames in conformity with the B format. However, the frames inconformity with both formats may contain samples for all channels and all sub- bands in thethird frame portion. A frame in conformity with the B format then lacks, in fact, theallocation information required to derive the samples for the channels I or II of the sub-bands17 to 32 from the third frame portion of a B format frame.

The receiver comprises a memory in which the allocation informationincluded in the second frame portion of an A format frame can be stored. If the next frame isa B format frame, only the allocation information for the sub-bands 1 to 16 and the channels Iand II in the memory is replaced by the allocation information included in the second frameportion of the B format frame. The samples for the sub-bands 17 to 32 from the third frameportion of the B format frame are derived from the allocation information for these sub-bandsderived from the preceding A format frame and still present in the memory. The reason forthe alternate use of A format frames and B format frames is that for some sub- bands, theallocation information (in the present case, the allocation information for the higher sub-bands17 to 32) does not change rapidly. Since, during quantization, the allocationinformation for the various sub-bands is available in the transmitter, this transmitter candecide to generate a B format frame instead of an A format frame if the allocationinformation for the sub-bands 17 to 32 inclusive does not change (significantly). Moreover,this illustrates that now additionalspace becomes available for the inclusion of samples in the third frame portion FD3.

For a specific value of P', the third frame portion of a B format frame is fourinformation packets longer than the third frame portion of an A format frame. This enablesthe number of bits by which the samples in thelower sub-bands 1 to 16 are represented, to beincreased, so that for these sub-bands, a higher transmission accuracy can be achieved.Moreover, if it is required to quantize the lower sub-bands more accurately, the transmitter can automatically opt for the generation of B format frames. This may then be at the expenseof the accuracy with which the higher sub-bands are quantized.

The third frame portion FD3 in Fig. 2 contains the samples of the quantizedsub-band signal components for the two channels. If theallocation word 0000 is not presentin the frame portion FD2 for any of the sub-band channels, this means that, in the presentexample, twelve samples are inserted in the third frame portion FD3 for each of the 32 sub-bandsand 2 channels. Thus, there are 768 samples in total.

SCALE FACTORS

In the transmitter, the samples may be multiplied by a scale factor prior totheir quantization. For each of the sub-bands and channels, the amplitudes of the twelvesamples are divided by the amplitude of that sample of the twelve samples, which has thelargest amplitude. In that case, a scale factor should be transmitted for every sub-band andevery channel in order to enable the inverse operation to be performed upon the samples atthe receiving end. For this purpose, the third frame portion then contains scale factors SF j,m,one for each of the quantized sub-band signal components in the various sub-bands.

In the present example, scale factors are represented by 6-bit numbers, themost significant bit first, the values ranging from 000000 to 111110. The scale factors of thesub-bands to which these are allocated, i.e., whose allocation information is non-zero, areaccommodated in the leading part of the frame portion FD3 before the samples. This meansthat the scale factors are transmitted before the transmission of the samples begins. Thisplacement of the scale factor information enables rapid decoding in thereceiver 5 to beachieved without the necessity of storing all the samples in the receiver, as will becomeapparent hereinafter. A scale factor SF j,m can thus represent the value by which the samplesof the signal in the j-th channel of the m-th sub-band have been multiplied. Conversely, thenumber one divided by this value may be stored as the scale factor so that, at the receivingend, it is not necessary to divide the scale factors before the samples are scaled up to thecorrect values.

For the frame format A, the maximum number of scale factors is 64. If theallocation word AW j,m for a specific channel j and a specific sub-band m has thevalue0000, which means that for this channel and this sub-band, no samples are present in theframe portion FD3, it will not be necessary to include a scale factor for this channel and thissub-band. The number of scale factors is then smaller than 64. The sequence in which thescale factors SF j,m are inserted in the third frame portion FD3 is the same as that in which the allocation words have been inserted in the second frame portion. The sequence istherefore as follows:

SF I,1; SF II,1; SF I,2; SF II,2; SF I,3; SF II,3; ...; SF I,32; SF II,32.

If it is not necessary to insert a scale factor, the sequence will not be complete.The sequence may then be, for example: ... SF I,4;SF 1,5; SF II,5; SF II,6;.... In this case, thescale factors for the fourth sub-band of channel II and the sixth sub-band of channel I are notinserted. If the frame is a B format frame, it may still be considered to insert scale factors inthe third frame portion for all the sub-bands and all the channels. However, this is not theonly option. In this case, it would also be possible to insert scale factors in the third frameportion of the frame for the sub-bands 1 to 16 only. In the receiver, this requires a memory inwhich all scale factors can be stored at theinstant at which a previously arriving A format frame is received. Subsequently, uponreception of the B format frame, only the scale factors for the sub-bands 1 to 16 are replacedby the scale factors included in the B format frame. The scale factors of the previouslyreceived A format frame for the sub-bands 17 to 32, are then used in order to restore thesamples for these sub-bands included in the third frame portion of the B format frame to thecorrect scale.

The samples are inserted in the third frame portion FD3 in the same sequenceas the allocation words and the scale factors, one sample for every sub-band of every channelin succession. According to this sequence, first, all the first samples for thequantized sub-band signals for all the sub-bands of both channels are inserted, then, all thesecond samples, ..., etc. The binary representation of the samples is arbitrary, the binary wordcomprising only "ones" preferably not being used again.

The transmission signal generated by thetransmitter 1 is subsequentlysupplied to thetransmission medium 4 by theoutput 7, and, by means of thetransmissionmedium 4, this signal is transferred to thereceiver 5. Transmission through thetransmissionmedium 4 may be a wireless transmission, such as, for example, a radio transmissionchannel. Many other transmission media are also possible. In this respect, opticaltransmission may be envisaged, for example, over optical fibers (real time) or optical recordcarriers (delayed time), such as Compact-Disc-like media, or transmission by means ofmagnetic record carriers utilizing RDAT or SDAT-like recording and reproducingtechnologies, for which reference is made to the book "The art of digital audio" by J.Watkinson, Focal Press, London 1988.

THE RECEIVER

As shown in Fig. 4, thereceiver 5 comprises a decoder, which decodes thesignal encoded in thecoder 6 of thetransmitter 1 and converts it into a replica of the wide-banddigital signal supplied to theoutput 8. The essential information in the incoming signalis contained in the scale factors and the samples. The remainder of the information in thetransmission signal is merely required for a "correct bookkeeping", to allow correct decoding.The receiver first derives the synchronizing and system information from the frames. Thedecoding process is then repeated for every incoming frame.

Fig. 12 shows a more detailed version of thereceiver 5 of Fig. 4. The codedsignal (the transmission signal) is applied through the terminal 10 to aswitch 11, aswitch 15and a synchronization andclock unit 19. For every frame, the synchronization andclock unit19 first detects the sync words situated in the first 16 bits of the first frame portion. Since thesync words of successive frames are, each time, spaced apart by an integral multiple of P' orP'+1 information packets, the sync words can be detected very accurately. Once the receiver isin synchronism, the sync word can be detected in the synchronization andclock unit 19. Toaccomplish this, a time window having, for example, a length of one information packet, isopened after each occurrence of P' information packets, so that only that part of the incominginformation is applied to the sync word detector in the synchronization andclock unit 19. Ifthe sync word is not detected, the time window remains open for the duration of anotherinformation packet, because the preceding frame may be aframe comprising P'+1 informationpackets. From these sync words, a PLL in the synchronization andclock unit 19 can derive aclock signal to control thecentral processing unit 18.

It is evident from the above that the receiver should know how manyinformation packets are contained in one frame. For this purpose, at the beginning of theframe, theswitch 15 is in the upper position shown, to apply the system information to theprocessing unit 18. The system information can now be stored in amemory 18a of theprocessing unit 18. The information relating to the number of information packets in a framecan be applied to the synchronization andclock unit 19 over a control-signal line 20, to openthe time window at the correct instants for sync-word detection. When the systeminformation is received, theswitch 15 is changed over to the lower position. The allocationinformation in the second frame portion of the frame can now be stored in thememory 18b.

If the allocation information in the incoming frame does not comprise anallocation word for all the sub-bands and channels, this will have become apparent alreadyfrom the detected system information. This may be, for example, the information indicating whether the frame is an A-format or a B-format frame. Thus, under the influence of therelevant information contained in the system information, theprocessing unit 18 will storethe received allocation words at the correct location in theallocation memory 18b.

It is obvious that in the present example, theallocation memory 18b comprises64 storage positions. If no scale factors are transmitted, the elements bearing thereferencenumerals 11, 12 and 17 may be dispensed with, and the content of the third frame portion of aframe is applied directly by a connection (not shown) from theinput 10 to asynthesis filter21. The samples are applied to thefilter 21 in the same sequence as the order in which thefilter 21 processes the samples in order to reconstruct the wide-band signal. The allocationinformation stored in thememory 18b is required in order to divide the serial data stream ofsamples into individual samples in thesynthesis filter 21, each sample having the correctnumber of bits. For this purpose, the allocation information is applied to thefilter 21 over theline 22.

The receiver further comprises ade-emphasis unit 23 which subjects thereconstructed digital signal supplied by thesynthesis filter 21 to de-emphasis. For a correctde-emphasis, the relevant information in thebits 24 to 31 of the first frame portion should beapplied from thememory 18a to thede-emphasis unit 23 over theline 24.

If the system uses scale factors in this format, the receiver will include theswitch 11, thememory 12, and themultiplier 17, and the third frame portion will contain thescale factors SF j,m. Because of a control signal applied by theprocessing unit 18 over theline 13, theswitch 11 is in the lower position at the instant at which the third frame portionFD3 of a frame arrives. Address signals are supplied to thememory 12 by theprocessing unit18 over theline 14. The scale factors are then stored in thememory 12, which has 64locations for the storage of the 64 scale factors. If a B-format frame is being received, theprocessing unit 18 applies such address signals to thememory 12 that only the scale factorsfor the sub-bands 1 to 16 are overwritten by the scale factors in the B-format frame.

Subsequently, as a result of another control signal applied over theline 13, theswitch 11 is changed to the upper position shown in the drawing, so that the samples areapplied to themultiplier 17. Using the allocation information, which is now applied to themultiplier 17 over theline 22, themultiplier 17 first derives the individual samples of thecorrect bit length from the serial data stream applied over theline 16. The samples are thenmultiplied so as to restore them to the correct values, which the original samples, had prior toscaling down in the transmitter. If the scale factors stored in thememory 12 are the scalefactor values by which the samples have been scaled down in the transmitter, these values should first be inverted (one divided by the value) before application to themultiplier 17.Obviously, it is also possible to invert the scale factors upon reception before they are storedin thememory 12. If the scale factors in the frames are already equal to the value by whichthe samples should be scaled up during reception, they can be stored directly in thememory12, and can then be applied directly to themultiplier 17.

It is evident that no memory is required to store all these samples beforestarting the signal processing performed upon the samples contained in the frame. At theinstant at which a sample arrives over theline 16, all the information required for processingthis sample is already available, so that processing can be carried out immediately. Thisentire process is controlled and synchronized by control signals and clock signals applied toall the parts of the transmitter by theprocessing unit 18.

Not all the control signals are shown. This is not necessary because the detailsof operation of the receiver will be obvious to those skilled in the art. Under control of theprocessing unit 18, themultiplier 17 multiplies the samples by the appropriate multiplicationfactors. The samples, which have now been restored to the correct amplitude, are applied tothereconstruction filter 21 in which the sub-band signals are reconverted to form the wide-banddigital signal. Further description of the receiver is not necessary because such receiversare generally known, for example, as described in the Thiele et al article cited above.Moreover, it will be evident that if the system information is also transmitted, the receivercan be highly flexible and can correctly decode the signals even if the transmission signalscontain different system information.

OTHER EMBODIMENTS

Fig. 13 shows, diagrammatically, another embodiment of the transmitter, inthe form of a recording device for recording the wide-band digital signal on a record carrier,such as amagnetic record carrier 25. Theencoder 6 supplies the transmission signal to arecording device 27 comprising awrite head 26, by means of which the signal is recorded ina track on the record carrier. It is then possible to record the transmission signal in a singletrack on the record carrier, for example, by means of a helical-scan recorder. In this case, thesingle track can be divided into juxtaposed tracks, which are inclined relative to thelongitudinal direction of the record carrier. An example of this is an RDAT-like recordingmethod. Another method is to split the information and simultaneously record the splitinformation in a plurality of juxtaposed tracks, which extend on the record carrier in thelongitudinal direction of the record carrier. For this, the use of an SDAT-like recording method may be considered. A comprehensive description of the two above methods can befound in the aforementioned book "The art of a digital audio" by J. Watkinson.

Again, it is to be noted that the signal supplied by theunit 6 may be first beencoded in a signal converter. This encoding may, for example, be an 8-to-10 conversionfollowed by an interleaving process, as described with reference to Fig. 4. If the transmissionsignal is recorded on the record carrier in a plurality of adjacent parallel track, the signalconverter should also be capable of assigning the encoded information to the various tracks.

Fig. 14 shows, diagrammatically, an embodiment of thereceiver 5, which maybe used in conjunction with the transmitter of Fig. 13; the two may form one apparatus,which then provides transmission over a period of time instead of distance. The receivershown is a player or read device for reading arecord carrier 25 according to the invention, onwhich the wide-band digital signal in the form of the transmission signal describedabove has been recorded by means of the device shown in Fig. 13. The transmission signal isread from a track on the record carrier by the readhead 29 and is applied to thereceiver 5,which may be, for example, of a construction as shown in Fig. 12. Again, theread device 28may be constructed to carry out an RDAT-like or an SDAT-like reproducing method. Bothmethods are described comprehensively in the aforementioned book by Watkinson.

If the signal supplied by theunit 6 in the recording device shown in Fig. 13has been converted, for example, in an 8- to-10 conversion and in an interleaving step, thetransmission signal read from therecord carrier 25 should first be de-interleaved and shouldbe subjected to 10-to-8 conversion. Moreover, if the transmission signal has been recorded ina plurality of parallel tracks, the reproducing unit shown in Fig. 14 should arrange theinformation read from these tracks in the correct sequence before further processing isapplied.

Figs. 15a-15d show a number of other possibilities of inserting the scalefactors and the samples in the third frame portion FD3 of a frame. Fig. 15a illustrates theabove-described method in which the scale factors SF for all the sub-bands m and channels (Ior II) are inserted in the third frame portion before the samples. Fig. 15b illustrates the samesituation as Fig. 15a, but in this case, it diagrammatically represents the storage capacity forthe scale factors SF I,m and SF II,m and the associated x samples for these two channels inthe sub-band m. Fig. 15b shows the samples for the two channels in the sub-band mcombined to blocks, whereas normally they are distributed within the third frame portion.The samples have a length of y bits. In the above example, x is 12 and y is now taken to be 8.

STEREO CODING

Fig.15c shows another format. The two scale factors for the first and thesecond channel in the sub-band are still present in the third frame portion. However, insteadof the x samples for both channels (the left and right channels for a stereo signal) in the sub-bandm (i.e., 2x samples in total), only x samples for the sub-band m are included in the thirdframe portion. These x samples are obtained, for example, by adding corresponding samplesin each of the two channels to one another. Thus, a monophonic signal is generated andtransmitted for this sub-band m.

The x samples in Fig. 15c each have a length of z bits. If z is equal to y, thissaves room in the third frame portion, which can be used for samples requiring a moreaccurate quantization. It is alternatively possible to express the x samples of the mono signalin Z = 2y (=16) bits. Such a signal processing is applied if the phase difference between theleft-hand and the right-hand signal components in a sub-band is irrelevant, but the waveformof the monophonic signal is important. This applies in particular to signals in higher sub-bandsbecause the phase-sensitivity of the ear for the frequency in these sub-bands is smaller.By expressing the x samples of the mono signal in 16 bits, the waveform is quantized moreaccurately, while the room occupied by these samples in the third frame portion is equal tothat in the example illustrated in Fig. 15b.

Yet another possibility is to represent the samples by an intermediate numberof bits, for example, 12 bits. The signal definition is then more accurate than in the exampleillustrated in Fig. 15b, while, at the same time, room is saved in the third frame portion sothat the bits saved can be allocated where the need is greater.

When the signals included in the third frame portion as illustrated in Fig. 15care reproduced at the receiving end, a stereo effect is obtained which is referred to as"intensity stereo". Here, only the intensities of the left-channel and the right-channel signals(in the sub-band m) can differ because of a different value for the scale factors SF I,m and SFII,m. Thus, different kinds of information relating to the stereo nature of the audio signal canbe represented by the composite signals and other signals, which are transmitted.

Fig. 15d shows still another possibility. In this case, there is only one scalefactor SFm for both signal components in the sub-band m. This is a situation, which isparticularly apt to occur in low-frequency sub-bands.

Yet another possibility, which is not shown, is that the x samples for thechannels I and II of the sub-band m, as in Fig. 15b, do not have associated scale factors SFI,m and SF II,m. Consequently, these scale factors are not inserted in the same third frame portion. In this case, the scale factors SF I,m and SF II,m included in the third frame portionof a preceding frame, must be used for scaling up the samples in the receiver.

All the possibilities described with reference to Figs. 15a-15d can beemployed in the transmitter in order to achieve a most efficient data transfer over thetransmission medium. Thus, frames as described with reference to different ones of Figs.15a-15d, may occur alternately in the data stream. It will be appreciated that, if the receiver isto be capable of correctly decoding these different frames, information about the structure ofthese frames must be included somewhere, such as in the system information.

THE TRANSMITTER

Fig. 16 shows the transmitter in more detail, particularly with respect tocombination of the various items of information to form the serial data stream shown in Figs.1, 2 and 3. Fig. 16 in fact shows a more detailed version of theencoder 6 in thetransmitter 1.Theencoder 6 has acentral processing unit 30, which controls a number of the encodercircuits, and also includes agenerator 31 for generating the synchronizinginformation and the system information described with reference to Fig. 3, agenerator 32 forsupplying allocation information, a generator 33 (optional) for supplying the scale factors, agenerator 34 for supplying the samples for a frame, and agenerator 35 for generating theadditional informationpacket IP P'+1.

The outputs of these generators are coupled to associated inputs of amultiplexer 40, shown as a five-position switch, whose output is coupled to theoutput 7 oftheencoder 6. TheCPU 30 controls the multiplexer (or switch) 40 over theline 53, and thevarious generators over the lines 41.1 to 41.4.

The operation of the transmitter will be described for a mono signal dividedinto M sub-band signals. These M sub-band signals S_SB1 to S_SBM are applied to the encoderinput terminals 45.1, 45.2, ..., 45.M. If scale factors are to be used, blocks of samples of eachof the sub-band signals are processed together in the optional sub-band scaling units 46.1 to46.M. A number, for example, twelve, of samples in a block are scaled to the amplitude ofthe largest sample in the block. The M scale factors are supplied to the unit 33 (if present)over the lines 47.1 to 47.M. The sub-band signals are supplied both to anallocation controlunit 49 and (scaled if that option is in use) to M quantizers 48.1 to 48.M. For every sub-band,theallocation control unit 49 defines the number of bits with which the relevant sub-bandsignal should be quantized. This allocation information is applied to the respective quantizers48.1 to 48.M over the lines 50.1 to 50.M, so that these quantizers correctly quantize the 12 samples of each of the sub-band signals, and is also supplied to thegenerator 32. Thequantized samples of the sub-band signals are supplied to thegenerator 34 over the lines 51.1to 51.M. The generators 32, 33 and 34 arrange the allocation information, the scale factorsand the samples in the correct sequence described above.

In the position of the multiplexer (or switch) 40 shown, the synchronizing andsystem information associated with the frame to be generated, is supplied by thegenerator 31in theCPU 30 and fed to theencoder output 7. Subsequently, the multiplexer (or switch) 40responds to a control signal supplied by theCPU 30 over theline 53, and is set to the secondposition from the top so that the output of thegenerator 32 is coupled to theoutput 7. Thesequence of allocation information is as described with reference to Fig. 10 or 11. After this,theswitch 40 is set to the third position from the top, coupling the output of thegenerator 33to theoutput 7, and thegenerator 33 now supplies the scale factors in the correct sequence.Theswitch 40 is then set to the next position, so that the output of thegenerator 34 is coupledto theoutput 7, and thegenerator 34 supplies the samples in the various sub-bands in thecorrect sequence. In this cycle, exactly one frame is applied to theoutput 7. Subsequently, theswitch 40 is reset to the top position. A new cycle is then started, in which a subsequent blockof 12 samples for each sub-band is encoded, and a subsequent frame can be generated on theoutput 7.

In some cases, for example, if the sample frequency F_S is 44.1 kHz (see Fig.5), an additional information packet (the dummy slot, see Fig. 2) must be added. In that case,after thegenerator 34 has finished supplying the samples, the multiplexer (or switch) 40 willbe set to the bottom position. The output of thegenerator 35 is now coupled to theoutput 7,and thegenerator 35 generates the additional informationpacket IP P'+1. After this, theswitch 40 is reset to the top position to start the next cycle.

It will be clear that, if the signal received by the transmitter is to be correctedfor errors caused during transmission of the signal, an appropriate error coding and/orinterleaving should be applied to the transmission signal. In addition, prior to transmission,some modulation (or channel encoding) is usually required. Thus, a transmission signaltransmitted through the transmission medium may not be directly identifiable as thetransmission signal, but will be a signal, which has been derived there from.

It will be noted that, for example, in the case that the sub-bands have differentwidths, the numbers of samples for the various sub-bands inserted in one third frame portionmay differ, and are likely to differ. If it is assumed, for example, that a division into three sub-bands is used, including a lower sub-band SB₁, a central sub-band SB₂and an upper sub-band SB₃, the upper sub-band may have a bandwidth which is, for example,twice as large as that of the other two sub-bands. This means that the number of samplesinserted in the third frame portion for the sub-band SB₃ is probably also twice as large as foreach of the other sub-bands.

The sequence in which the samples are applied to the reconstruction filter inthe decoder may then be: the first sample of SB₁, the first sample of SB₃, the first sample ofSB₂, the second sample of SB₃, the second sample of SB₁, the third sample of SB3, thesecond sample of SB₂, the fourth sample of SB₃,..., etc. The sequence in which the allocationinformation for these sub-bands is then inserted in the second frame portion will then be:first, the allocation word for SB₁, then, the allocation word of SB₃, and subsequently, theallocation word for SB₂. The same applies to the scale factors. Moreover, the receiver canderive, from the transmitted system information, that, in this case, the cycle comprises groupsof four samples each, each group comprising one sample of SB₁, one sample of SB₃, onesample of SB₂ and subsequently, another sample of SB₃.

OTHER FRAME ARRANGEMENTS

Figure 17 shows another structure of the first frameportion FD1. Again, the first frame portion FD1 contains exactly 32 bits and, therefore,corresponds to one information packet. The first 16 bits again constitute the synchronizingsignal (or synchronization word). The synchronization word may also be the same as thesynchronization word of the first frame portion FD1 in Fig. 3, but the Fig. 17 informationaccommodated inbits 16 through 31 differs from the information inbits 16 through 31 inFig. 3. The bits b₁₆ through b₁₉ represent a 4-bit bit rate index (BR index) number whosemeaning is illustrated in the table in Fig. 18. If the bit rate index is equal to the 4-bit digitalnumber '0000', this denotes the free-format condition, which means that the bit rate is notspecified and that the decoder has to depend upon the synchronization word alone to detectthe beginning of a new frame. The 4-bit digital number '1111' is not employed in order not todisturb the synchronization word detection. In the second column of the table of Fig. 18, thebit rate index is represented as a decimal number corresponding to the 4-bit digital number.The corresponding bit rate values are given incolumn 1.

With this format, the first frame portion contains information related to thenumber of information packets in the frame. As shown in Fig. 18, the sample frequency F_S isdefined by one of the four possible 2-bit digital numbers for the bits b₂₀ and b₂₁ having the values listed.Bit 22 indicates whether the frame comprises a dummy slot, in which case b₂₂ ='1', or does not comprise a dummy slot, in which case b₂₂ = '0'. Along with otherpredetermined information, then, the information in the bits b₁₆ through b₂₂ makes it possibleto determine how many information packets are actually present in the frame.

From the number of samples of the wide-band signal whose correspondinginformation belonging to the transmission signal is accommodated in one frame, in thepresent example, n_S = 384, it is possible to determine how many information packets B arepresent in the frame by means of the data in the table in Fig. 8, the padding bit b₂₂ and theformulaP = (BR x nS) / (N x FS).The bit b₂₃ is intended for specifying a future extension of the system. This future extensionwill be described hereinafter. For the time being, this bit is assumed to be '0'.

INDICATOR SIGNALS

Various indicator and control signals are provided by the bits b₂₄ through b₃₁,which will be described with reference to Figs. 19 and 20. The bits b₂₄ and b₂₅ give the modeindication for the audio signal. For the four possibilities of this two-bit digital number, Fig.20 shows whether the wide-band digital signal is a stereo audio signal ('00'), a mono signal('11'), a bilingual signal ('10'), or an intensity stereo audio signal ('01'). In the last-mentionedcase, thebits 26 and 27 indicate which sub-bands have been processed in accordance with theintensity stereo method. In this example, the respective two-bit numbers '00', '01', '10', and'11' mean, respectively, that the sub-bands 5-32, 9-32, 13-32 and 17-32 have been processedin accordance with the intensity stereo method. As stated hereinbefore, intensity stereo can beapplied to the higher sub-bands because the ear is less phase- sensitive for the frequencies inthese sub-bands.

The bit b₂₈ can be used as a copyright bit. If this bit is '1', this means that theinformation is copy-protected and should/cannot be copied. The bit b₂₉ can indicate that theinformation is original information (b₂₉ = '1'), for example, in the case of prerecorded tapes,or information, which has been copied (b₂₉ = '0'). The bits b₃₀ and b₃₁ specify the emphasis,which may have been applied to the wide-band signal in the transmitter, for example, asdescribed with reference to Fig. 7.

Various configurations of the second frame portion FD2 may be described bythe various mode indications represented by the bits b₂₄ through b₂₇ in the first frame portion.The second frame portion comprises the 4-bit allocation words whose meaning has been described with reference to Fig. 9. For the stereo mode (b₂₄, b₂₅ = 00) and the bilingual mode(b₂₄, b₂₅ = 10), the second frame portion FD2 again has a length of 8 information packets(slots) and is composed as described with reference to Fig. 10. In the stereomode, 'I' in Fig. 10 then represents, for example, the left-channel component and 'II'represents the right channel component. For the bilingual mode, 'I' denotes one language and'II' denotes the other language. For the mono mode (b₂₄, b₂₅ = 11), the length of the secondframe portion FD2 is, of course, only 4 information packets (slots).

Fig. 21 illustrates the sequence of the allocation words for thevarious sub-bands1 through 32 in the four information packets (slots) 2 through 5. Thus, every quantityM-i represents a four-bit allocation word, which specifies the number of bits in every samplein the sub-band of the sequence number i, i ranging from 1 to 32. In the intensity stereo mode(b₂₄, b₂₅ = 01), there are four possibilities indicated by means of the bits b₂₆ and b₂₇, seeFig. 20. All of these possibilities result in a different content of the second frame portionFD2.

Figs. 22a-22d illustrate the four different contents of the second frame portion.If the switch bits b₂₆, b₂₇ are '00', the signals in thesub-bands 1 through 4 are normal stereosignals and the signals in thesub-bands 5 through 32 are intensity-stereo signals. This meansthat for the sub-bands 1 through 4, for the left-hand and right-hand channel components inthese sub-bands, the associated allocation words should be stored in the second frameportion. In Fig. 22a, this is represented by the consecutive allocation words AW (L, 1); AW(R, 1); AW (L, 2); AW (R, 2); ...; AW (R, 4), stored in theslot 2 of the frame, i.e., the firstslot of the second frame portion. Fig. 22a only gives the indices (i-j) of the allocation words, ibeing equal to L or R and indicating the left-hand and the right-hand channel components,respectively, and j ranging from 1 through 4 and representing the sequence number of thesub-band. For thesub-bands 5 through 32, the left-hand and the right-hand channelcomponents contain the same series of samples. The only difference resides in the scalefactors for the left-hand and the right-hand channel components in a sub-band. Consequently,such a sub-band requires only one allocation word. The allocation words AW (i, j) for thesesub-bands 5 through 32 are indicated by the indices M-j, where i is consequently equal to Mfor all the sub-bands and where j ranges from 5 through 32.

Fig. 22a shows that 4-1/2 information packets are required for inserting the 36allocation words in the second frame portion. If the switch bits b₂₆, b₂₇ are '01', the signals inthesub-bands 1 through 8 will be normal stereo signals and the signals in thesub-bands 9 through 32 will be intensity-stereo signals. This means that for each of the sub-bands 1through 8, two allocation words AW(L, j) and AW(R, j) are required and that foreach of the sub-bands 9 through 32, only one allocation word AW(M, j) is required. Thisimplies that, in total, 40 allocation words are needed, included in five information packets(slots), i.e., IP2 through IP6, of the frame. This is illustrated in Fig. 22b. In this case, thesecond frame portion FD2 has a length of five information packets (slots).

If the switch bits b₂₆, b₂₇ are '10', the signals in thesub-bands 1 through 12 willbe normal stereo signals and the signals in the sub-bands 13 through 32 will be intensity-stereosignals. Fig. 22c gives the structure of the second frame portion FD2 with theallocation words for the various sub-bands. The second frame portion now has a length of 5-1/2information packets (slots) in order to accommodate all the allocation words. If theswitch bits b₂₆, b₂₇ are '11', the signals in thesub-bands 1 through 16
will be normal stereo signals and the signals in the sub-bands 17 through 32 will be intensity-stereosignals. Now, 48 allocation words are needed, which are inserted in the second frameportion, which then has a length of 6 information packets (slots), see Fig. 22d.

What has been stated above about the scale factors is also valid here. When itis assumed that anallocation word 0000 has been assigned neither to any of the sub-bandsnor to any of the channels, 64 scale factors are required both for the stereo mode and for theintensity-stereo modes. This is because in all the intensity-stereo modes, every mono sub-bandshould have two scale factors to enable intensity-stereo to be realized for the left-handand the right-hand channel in this sub-band (see Fig. 15c). It is obvious that in the monomode, the number of scale factors is halved, i.e., 32, again assuming that theallocation word0000 has not been assigned to any of the sub-bands.

SCALE FACTOR DETERMINATION

A method of determining the 6-bit scale factors will now be explained below.As stated hereinbefore, the sample having the largest absolute value is determined for every12 samples of a sub- band channel. Line (a) of Fig. 24 shows the binary representation of amaximal sample |S_max|. The first bit, designated SGN, is the sign bit and is '0' because itrelates to the absolute value of S_max. The samples are represented in two's complementnotation. The sample comprises k 'zeros' followed by a "1". The values of the other bits of the24-bit digital number are not relevant and can be either '0' or '1'.

|S_max| is now multiplied by 2k to produce the number shown in line (b) of Fig.24. Subsequently, |S_max|·2^k is compared with a digital number DV₁ equal to 010100001100000000000000 and a digital number DV₂ equal to011001100000000000000000. If |S_max|·2^k < DV₁, a specific constant p is taken to be 2. IfDV₁ ≤ |S_max|·2^k < DV₂, then p is taken to be 1. If |S_max|·2^k ≥ DV₂, then p=O.

The number k is limited to 0 ≤ k ≤ 20. The scale factor is now determined bythe numbers k and p in accordance with the following formula:SF = 3k + p.Consequently, the maximum value for SF is 62. This means that the scale factors can berepresented by 6-bit numbers, the six-bit number 111111 (which corresponds to the decimalnumber 63) not being used. In fact, the 6-bit binary numbers are not the scale factors, butthey are in a uniquely defined relationship with the actual scale factors, as will be set forthbelow. All of the 12 samples S are now multiplied by a number, which is related to the valuesfor k and p. The 12 samples are each multiplied as follows:S'= S x 2k x g(p)where the number g(p) has the following relation with p:g(p) = 1 for p = 0g(p) = 1+2-2+2-8+2-10+2-16+2-18+2-23 for p = 1g(p) = 1+2-1+2-4+2-6+2-8+2-9+2-10+2-13+2-15+2-16+2-17+2-19+2-20 for p = 2.

The parameter k specifies the number of 6 dB steps and the factors g(1) andg(2) are the closest approximations to steps of 2 dB. The samples S' thus scaled are nowquantized to enable them to be represented by q-bit digital numbers in two's complementnotation. In Fig. 25, this is illustrated for q = 3. The scaled samples S' have values between+1 and -1, see Fig. 25a. In the quantizer, these samples must be represented by q bits, qcorresponding to the allocation value for the relevant sub-band (channel). Since, as statedabove, the q-bit digital number comprising only 'ones' is not used to represent a sample, thetotal interval from -1 to +1 should be divided over 2^q-1 smaller intervals. For this purpose, thescaled samples S' are transformed into the samples S" in accordance with the formula:S"= S' (1-2-q)-2-q.

The samples S" are subsequently truncated at q bits, see Fig. 25c. Since the'111' representation is not permissible, the sign bits are inverted, see Fig. 25d. The q(=3)-bitnumbers given in Fig. 25d are now inserted in the third frame portion FD3, see Fig. 2.

Samples S' which comply with -0.71 ≤ S' ≤ -0.14 are represented by the digitalnumber '001'. This proceeds similarly for samples S' of larger values up to samples which comply with 0.71 ≤ S' < 1, and which are represented by the digital number '110'.Consequently, the digital number '111' is not used.

Dequantization at the receiving side is effected in a manner inverse to thequantization at the transmission side, see Fig. 26. This means that first, the sign bits of the q-bitdigital numbers are inverted to obtain the normal two's complement notation, see Fig. 26b.

Subsequently, the samples S' are derived from the transformed samples S" bymeans of the formula:S' = (S"+ 2-q+1) (1 + 2-q + 2-2q + 2-3q + 2-4q + ...)(see Figs. 26c and 26d). The values S' thus obtained are now situated exactly within theoriginal intervals in Fig. 25a. At the receiving side, the samples S' are subsequently scaled tothe original amplitudes by means of the transmitted information k, p which is related to thescale factors. Thus, at the receiving side, a number g'(p) complies with:g'(p) = 1 for p = 0g'(p) = 2-1 + 2-2 + 2-5 + 2-6 for p = 1g'(p) = 2-1 + 2-3 + 2-8 + 2-9 for p = 2.Scaling to the original amplitudes is now effected using the following formula:S = S'·2-k·g'(p).

In the two possible versions of a frame as described with reference to Figures2 and 3 and Figures 2, 17 and 19, respectively, the third frame portion may not be filledentirely with information. This will occur more often and sooner as the algorithms for sub-bandcoding, i.e., the entire process of dividing the signal into sub-band signals and thesubsequent quantization of the samples in the various sub-bands, are improved. In particular,this will enable the information to be transmitted with a smaller number of bits (averagenumber per sample). The unused part of the third frame portion can then be utilized fortransmitting additional information. In the first frame portion FD1 in Fig. 17, allowance hasbeen made for this by means of the "future-use" bit b₂₃. Normally, this bit is '0', as will beapparent from Fig. 18.

ADDITIONAL SIGNAL

If an additional signal has been inserted in the third frame portion FD3 of aframe, the future-use bit b₂₃ in the first frame portion FD1, see Fig. 17, will be '1'. Duringreading of the first frame portion FD1, this makes it possible for the receiver to detectwhether the frame contains additional information. The allocation information and the scalefactors, see Fig. 23, inform the receiver that only the part of the third frame portion FD3, marked FD4 in Fig. 23, contains quantized samples of the sub-band signals. The remainder,marked FD5 in Fig. 23, now contains the additional information. The first bits in this frameportion FD5 are designated 'EXT INFO' or extension information. These bits indicate thetype of additional information. The additional information may be, for example, an additionalaudio channel, for example, for the transmission of a second stereo channel. Anotherpossibility is to use these two additional audio channels to realize 'surround sound' togetherwith the audio sub-band signals in the frame portion FD4. In that case, the front-rearinformation required for surround sound may be included in the frame portion FD5. In thepart marked FD6, the frame portion FD5 may again contain allocation information, scalefactors and samples (in this order), and the sequence of the allocation words and the scalefactors may then be similar to the sequence as described with reference to Figs. 2 and 3 andFigs. 2, 17 and 19.

In the case of 'surround sound', simple receivers may merely decode the stereoaudio information in the frame portions FD2 and FD3, except for the frame portion FD5.More sophisticated receivers are then capable of reproducing the surround-sound informationand, for this purpose, they also employ the information in the frame portion FD5.

The extension-info bits may also indicate that the information in the frameportion FD6 relates to text, for example, in the form of ASCII characters. It may even beconsidered to insert video or picture information in the frame portion FD6, this informationagain being characterized by the extension-info bits.

It is to be noted that the invention is not limited to the embodiments shownherein. The invention also relates to those embodiments, which differ from the embodimentsshown herein with respect to features, which are not relevant to the invention as defined inthe claims.

Claims (14)

1. A record carrier (25) having an encoded wide-band digital audio signalrecorded on it,
   the wide-band digital audio signal comprising at least a first and a second signal component,the signal components being filtered into sub signals for the at least two signal components,
   a sub signal comprising sample information,
   characterized in that from at least one corresponding sub signal of the at least first and secondsignal components a composite sub signal may have been derived,
   the record carrier having recorded on it sub signals of the at least two signal components thathave not been processed into a composite sub signal, further a mode indicator control signal(b₂₄,b₂₅) indicating whether a corresponding sub signal of the at least two signal componentshave or have not been processed into a composite sub signal also being recorded on therecord carrier.
2. Record carrier as claimed in claim 1,characterized in that further a sub signalindicator control signal (b₂₆,b₂₇) indicating which at least one corresponding sub signal of theat least two signal components have been processed also being recorded on the record carrier.
3. Record carrier as claimed in claim 1 or 2,characterized in that, if the modeindicator signal indicates that a corresponding sub signal of the at least two signalcomponents have been processed, the composite sub signal obtained from processing saidcorresponding sub signals of the at least two signal components also being recorded on therecord carrier.
4. Record carrier as claimed in claim 1 or 2, a sub signal comprising scale factorinformation in addition to the sample information,
   characterized in that, if the mode indicator signal indicates that a corresponding sub signal ofthe at least two signal components have been processed into a composite sub signal,
   the composite sub signal obtained from processing said corresponding sub signals of the atleast two signal components also being recorded on the record carrier,
   as well as information representative of the respective scale factors of all the sub signals ofthe at least two signal components.
5. Record carrier as claimed in claim 1,characterized in that the mode indicatorsignal is a 2-bit signal,
   where the bit combination '01' indicates that the corresponding sub signal of the at least twosignal components have been processed.
6. Record carrier as claimed in claim 5,characterized in that the bit combination'00' indicates that no sub signals have been processed into composite sub signals.
7. Record carrier as claimed in claim 2,characterized in that the sub signalindicator control signal (b₂₆,b₂₇) indicates which m₁ corresponding sub signals of the at leasttwo signal components have been processed, where m₁ is larger than 1.
8. Record carrier as claimed in claim 7,characterized in that m₁ can vary overtime.
9. Record carrier as claimed in claim 7 or 8, the m₁ sub signals are the signalscorresponding to the highest frequency bands in the frequency range of the wideband digitalaudio signal.
10. Record carrier as claimed in claim 9,characterized in that the sub signalindicator control signal (b₂₆,b₂₇) is a 2-bit signal.
11. Record carrier as claimed in claim 10,characterized in that the bit combination'11' indicates a predetermined value for m₁, the bit combination '10' indicates a value for m₁which equals the value for m₁ with the bit combination '11' plus 4.
12. Record carrier as claimed in claim 11,characterized in that the bit combination'01' indicates a value for m₁ which is the value for m₁ with the bit combination '11' plus 8.
13. Record carrier as claimed in claim 12,characterized in that the bit combination'00' indicates a value for m₁ which is the value for m₁ with the bit combination '11' plus 12.
14. Record carrier as claimed in claim 11,characterized in that the value for m₁with the bit combination '11' is 16.

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