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Claims  |
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What is claimed is:
1. A digital transmission system, for producing a replica of a digital signal comprising at least a first component and a second component, comprising:
an encoder including analysis means for altering said digital signal to obtain a plurality of sub-signals, including at least a first sub-signal and a second sub-signal from said first component and said second component respectively,
signal combination means for combining said first sub-signal and said second sub-signal to obtain a composite sub-signal,
signal generator means for generating an indicator signal indicating that said first sub-signal and said second sub-signal are combined,
transmission means for transmitting said indicator signal, said composite sub-signal, and sub-signals which have not been combined,
receiving means for receiving the signals which were transmitted,
means, responsive to the received indicator signal and composite subsignal, for generating a signal related to at least one of said first component and said second component, and
a decoder including synthesis means for combining the transmitted subsignals and the signal related to at least one of said first component and said second component to produce said replica of the digital signal.
2. A system as claimed in claim 1, characterized in that said first and second component represent information relating to the stereo nature of an audio signal.
3. A system as claimed in claim 1, characterized in that said analysis means subdivides said digital signal into sub-signals which are subband signals representing respective frequency subbands and said components.
4. A digital transmission system, for producing a replica of a digital signal comprising at least a first and a second component, comprising a transmitter and a receiver,
wherein said transmitter comprises:
an encoder including analysis means for filtering said digital signal to obtain subband signals for M subbands, where M>1, said subband signals including a plurality of first subband signals and a plurality of second subband signals from said
first component and second component respectively,
signal combination means for combining m.sub.1 of said first subband signals respectively with m.sub.1 of said second subband signals from corresponding subbands to obtain m.sub.1 composite subband signals, where 1<m.sub.1 <M,
signal generator means for generating an indicator signal indicating which said first and second subband signals are combined, and
transmission means for transmitting said indicator signal, said composite subband signals, and subband signals which were not combined, and
the receiver comprises:
receiving means for receiving the signals which have been transmitted,
detection means for detecting said indicator signal,
derivation means responsive to the received indicator signal, for producing, from the received composite subband signals, derived subband signals related to m.sub.1 of said first subband signals and m.sub.1 of said second subband signals, and
a decoder including synthesis means for combining said derived subband signals and the received subband signals which were not combined, to produce said replica of the digital signal.
5. A system as claimed in claim 4, wherein said analysis means applies substantially identical filtering to said first and second components to obtain said first and second subbands.
6. A system as claimed in claim 4, characterized in that said m.sub.1 of the first subband signals are subband signals for the m.sub.1 highest frequency subbands.
7. A system as claimed in claim 4, characterized in that said digital signal represents a first block of samples and a second block of samples, said first component and said second component being first block first and second components, said
subband signals being first block subband signals, said m.sub.1 of said first and second subband signals being m.sub.1 of the first block first and second subband signals, said m.sub.1 composite subband signals being m.sub.1 first block composite subband
signals, and said replica being a replica of the portion of said digital signal representing said first block of samples,
for producing a replica of the portion of said digital signals representing said second block of samples, said analysis means obtains second block subband signals for said M subbands including corresponding pluralities of second block first and
second subband signals from second block first and second components respectively,
said signal combination means combines in each of a number m.sub.2 subbands the second block subband signals from the respective second block first and second components, to obtain m.sub.2 composite signals in said m.sub.2 subbands, where m.sub.2
is greater than m.sub.1,
said signal generator means generates a second block indicator signal identifying said m.sub.2 subbands,
said transmitter transmits said composite signals in said m.sub.2 subbands, said second block indicator signal, and second block subband signals which were not combined, and
said derivation means derives said m.sub.2 composite signals in said m.sub.2 subbands from the second block signal received, and derives from said m.sub.2 composite signals in said m.sub.2 subbands, in response to said second block indicator
signal, subband signals in said m.sub.2 subbands corresponding to said second block subband signals which were combined.
8. A system as claimed in claim 7, characterized in that said m.sub.2 subbands are the m.sub.2 highest subbands.
9. A system as claimed in claim 6, characterized in that said first and second components are respective stereo audio signals.
10. A system as claimed in claim 4, characterized in that said transmitter comprises a scale factor determiner, for determining a scale factor for time equivalent subband signal blocks of the first and second components in the subband signals;
and means for transmitting these scale factors, and
said detector in the receiver is adapted to detect the scale factors which have been transmitted.
11. A system as claimed in claim 2, characterized in that said transmitter comprises means for quantizing the time equivalent signal blocks of the subband signals and the one or more composite signals.
12. A transmitter, for transmitting signals representative of a digital signal comprising at least a first component and a second component, over a transmission medium, comprising:
an encoder including analysis means for altering said digital signal to obtain a plurality of sub-signals, including at least a first sub-signal and a second sub-signal from said first component and said second component respectively,
signal combination means for combining said first sub-signal and said second sub-signal to obtain a composite sub-signal,
signal generator means for generating an indicator signal indicating that said first sub-signal and said second sub-signal are combined,
transmission means for transmitting said indicator signal, said composite sub-signal, and sub-signals which have not been combined.
13. A transmitter as claimed in claim 12, characterized in that said first and second components are respective stereo audio signals.
14. A transmitter as claimed in claim 12, characterized in that said analysis means filters said digital signal to provide sub-signals which are subband signals representing said digital signal in M respective frequency subbands, where M>1,
said subband signals including a plurality of first subband signals and a plurality of second subband signals from said first component and second component respectively,
said signal combination means combines m.sub.1 of said first subband signals respectively with m.sub.1 of said second subband signals from corresponding subbands to obtain m.sub.1 composite subband signals, where 1<m.sub.1 <M, and
said indicator signal indicates which said first and second subband signals are combined.
15. A transmitter as claimed in claim 14, characterized in that said digital signal represents a first block of samples and a second block of samples, said first component and said second component being first block first and second components,
said subband signals being first block subband signals, said m.sub.1 of said first and second subband signals being m.sub.1 of the first block first and second subband signals, said m.sub.1 composite subband signals being m.sub.1 first block composite
subband signals, and said replica being a replica of the portion of said digital signal representing said first block of samples,
for producing a replica of the portion of said digital signals representing said second block of samples, said analysis means obtains second block subband signals for said M subbands including corresponding pluralities of second block first and
second subband signals from second block first and second components respectively,
said signal combination means combines in each of a number m.sub.2 subbands the second block subband signals from the respective second block first and second components, to obtain m.sub.2 composite signals in said m.sub.2 subbands, where m.sub.1
<m.sub.2 .ltoreq.M,
said signal generator means generates a second block indicator signal identifying said m.sub.2 subbands, and
said transmission means transmits said m.sub.2 composite signals, said second block indicator signal, and second block subband signals which were not combined.
16. A transmitter as claimed in claim 14, wherein said analysis means applies substantially identical filtering to said first and second components to obtain said first and second subbands.
17. A transmitter as claimed in claim 14, characterized in that said m.sub.1 of the first subband signals are subband signals for the m.sub.1 highest frequency subbands.
18. A transmitter as claimed in claim 14, characterized in that said transmitter comprises a scale factor determiner, for determining a scale factor for time equivalent subband signal blocks of the first and second components in the subband
signals; and means for transmitting these scale factors.
19. A receiver for producing a replica of a digital signal including a first component and a second component, from digital signal components comprising at least one composite sub-signal, an indicator signal indicating that at least a first and
a second sub-signal are combined, and a plurality of subsignals not including said first and second sub-signal, said digital signal components being representative of said digital signal,
receiving means for receiving said digital signal components,
means, responsive to the received indicator signal and composite subsignal, for generating a signal related to at least one of said first component and said second component, and
a decoder including synthesis means for combining the transmitted subsignals and the signal related to at least one of said first component and said second component to produce said replica of the digital signal.
20. A receiver as claimed in claim 19, characterized in that said first and second components are respective stereo audio signals.
21. A receiver as claimed in claim 19, characterized in that said composite sub-signal represents a first subband signal and a second subband signal for a combined subband, and said sub-signals are subband signals representing respective
frequency subbands other than said combined subband, and
said means for generating comprises derivation means for deriving said composite subsignal from the signal received and for deriving from said composite signal, in response to the indicator signal, subband signals corresponding to said first
component and said second component.
22. A receiver as claimed in claim 21, characterized in that the digital signal components comprise a plurality of said composite sub-signals representing a plurality of combined subbands respectively, and a scale factor for time equivalent
signal blocks of the first component and the second component,
said detector in the receiver is adapted to detect said scale factor, and
said derivation means is responsive to said scale factor. |
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Description |
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BACKGROUND OF THE INVENTION
1. Field of the Invention
The invention relates to a transmission system for producing a replica of a wideband digital signal which includes at least a first and a second component; and more particularly to such a system which comprises an encoder including an analyzer
for altering the digital signal to obtain a number n of sub-signals for said digital signal; a transmitter for transmitting the sub-signals for reception at a different time or place; a receiver for receiving the sub-signals; and a decoder including a
synthesizer for combining the received sub-signals to obtain respective replicas of the digital signal. The invention also relates to an encoding transmitter, and a decoding receiver for such a system.
2. Description of the Prior Art
One transmission system of this type is known from an article, "The Critical Band Coder-Digital Encoding of Speech Signals Based on the Perceptual Requirement of the Auditory System" by M. E. Krasner, published in Proc. IEEE ICASSP 80, Vol. 1,
pp 327-331, Apr. 9-11, 1980. This article relates to a transmission system in which the sub-signals are signals representing frequency bands. The transmitter includes a frequency subband coding system in which the speech signal band is divided into a
plurality of subbands whose bandwidths approximate the bandwidths of the critical bands of the human ear in the respective frequency ranges (see FIG. 2 of this article). This division is selected because, based on psycho-acoustic experiments, one can
expect that quantization noise in such a subband will be masked to an optimum extent by the signals in that subband, if during quantization allowance is made for the noise-masking curve of the human ear. Threshold values for noise masking by m single
tones are shown in FIG. 3 of this article. The receiver employs a corresponding subband decoding system.
When applying frequency subband coding to a high-quality digital music signal, such as one according to the Compact Disc Standard which uses 16 bits per signal sample at a sample frequency of 1/T=44.1 kHz, with a suitably selected bandwidth and a
suitable selected quantization for the respective subbbands, the quantized output signals of the coder can be represented by an average number of approximately 2.5 bits per signal sample. The quality of the replica of the music signal does not differ
perceptibly from that of the original music signal in substantially all passages of substantially all kinds of music signals.
The subbands need not necessarily correspond to the bandwidths of the critical bands of the human ear. For example, the subbands may have equal bandwidths, provided that allowance is made for this in determining the masking threshold.
The invention is also applicable to other types of transmission systems, such as those in which blocks of samples are transform coded. Such systems are referred to in the article "Low bit-rate coding of high-quality audio signals. An
introduction to the MASCAM system" by G. Thiele, G. Stoll and M. Link, published in EBU Technical Review, no. 230, pp. 71-94, August 1988. In such a system the transform coefficients correspond to the sub-signals.
The sub-signal transmission systems described above have the disadvantage that, in some cases, perceptible differences occur between the replica and the signal which was to be transmitted. These differences are perceived as a form of distortion
in the replica generated by the receiver. Often they are the result of the number of bits, available for quantization of certain of the sub-signals, being too low.
SUMMARY OF THE INVENTION
An object of the invention is to enable transmission of signals representing a wideband digital signal with a significant reduction of the distortion present in the replica generated in the receiver.
Another object of the invention is to identify sub-signals, corresponding to first and second signal components which are related to each other, which can be combined to obtain a composite signal which is transmitted in place of those
sub-signals.
Another object of the invention is to transmit digital signals corresponding to stereo audio signals with reduced distortion in the generated replica.
According to the invention, a system as described above further comprises control circuitry for determining and optimizing bit allocation, and a signal combiner for combining selected corresponding sub-signals from the first and second components
of the original digital signal to obtain one or more composite sub-signals, and an indicator generator for generating an indicator signal indicating that these corresponding sub-signals are combined. The receiver is responsive to the indicator signal,
for generating a signal relating to that composite signal and related to at least one of said first and second components. Preferably, the receiver includes a decoder which synthesizes a signal which is the desired replica, by combining the transmitted
sub-signals and composite sub-signals.
The invention is based on recognition that the numbers of bits made available for different sub-signals are not optimally allocated, so that quantization of certain sub-signals is too rough. This leads to audible distortion in a replica
resulting from decoding of the received signal. By selectively combining subsignals which have a correspondence or relationship to each other, and quantizing only one composite sub-signal, so as to make more bits available for quantizing of those
sub-signals which are transmitted, the reduced quantizing distortion may more than compensate for the slight loss of information in the replica. This is especially true when the sub-signals which are combined are signals corresponding to a same
frequency subband, such as left and right stereo, or other spatial-differentiating signals, in music or audio transmission.
Alternatively, the composite signal may itself be quantized with a greater number of bits than if the two sub-signals were quantized separately.
In a preferred embodiment, a control or central processing unit, and an allocation control unit, together functioning as control or steering circuits, control the signal combiner to combine in each of a number m.sub.1 of said subbands the subband
signals of the first and second components in those subbands, to obtain m.sub.1, composite signals in said m.sub.1 subbands, where m.sub.1 is greater than 1. The signal generator generates an indicator signal identifying which subbands had their
corresponding sub-signals combined. This indicator signal will function as a steering control signal. The transmitter transmits these composite signals in the m.sub.1 subbands, the indicator signal, and the remaining sub-signals which have not been
combined. In this embodiment, the receiver decoder has a deriving circuit for deriving m.sub.1 subband signals from the composite signals in the m.sub.1 subbands, and combining these with the subband signals which were transmitted.
A variation of this embodiment allows a still greater reduction of the data. An allocation control circuit in the transmitter determines the bit availability after the m.sub.1 subbands have been processed to form the composite sub-signals. If
bit availability is still such that quantization of some subbands will be too rough, then a second evaluation is made with a greater number of subbands being combined. For example, in each of a number m.sub.2 subbands the subband signals of said signal
components are combined to obtain m.sub.2 composite signals in said m.sub.2 subbands. The value m.sub.2 will be greater than m.sub.1, and will preferably include all of the m.sub.1 subbands.
In this embodiment the signal generator will then generate a different indicator signal, identifying the m.sub.2 subbands, and the transmitter transmits these composite signals in the m.sub.2 subbands. In the receiver the deriving circuit
derives first the m.sub.2 composite signals in the m.sub.2 subbands from the signal received, and then derives from these m.sub.2 composite signals, in response to the indicator signal, subband signals in the m.sub.2 subbands corresponding to said signal
portions.
In a typical audio subband division, the m.sub.1 subbands are the m.sub.1 highest subbands; and if further combining is required, then the m.sub.2 subbands are the m.sub.2 highest subbands. This method of combining takes advantage of the fact
that the human ear is less phase sensitive in those frequency bands. In one embodiment discussed more fully below, the value m.sub.1 is half of the number M of subbands. For example, if M=32, the highest (highest frequency) 16 bands may initially be
selected for combining. A value of 20 may be used for m.sub.2, and the process can be repeated for m.sub.3 =24 and m.sub.4 =28.
In yet another preferred embodiment, the transmitter comprises a scale factor determiner, for determining a scale factor for time equivalent signal blocks of the first and second components in the subband signals; and the transmitting section
transmits these scale factors. The detector in the receiver is adapted to detect the scale factors which have been transmitted, and to control a multiplier for the subband signals before the full bandwidth signal is reconstructed in a synthesis filter.
Correction for any pre-emphasis is made after reconstruction.
BRIEF DESCRIPTION OF THE DRAWING
FIG. 1 is a diagram of a second digital signal generated by a transmitter according to the invention, organized as frames each composed of information packets,
FIG. 2 is a diagram of the structure of a frame according to a preferred embodiment 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 system according to the invention
FIG. 5 is a table showing the number of information packets B in a frame, for certain values of bit rate BR and sample frequency F.sub.s,
FIG. 6 is a table showing the numbers of frames in a padding sequence for different bit rates,
FIG. 7 is a table showing the system information included in the first portion of a frame,
FIG. 8 is a table showing a distribution of information between channels for diffferent modes,
FIG. 9 is a table of meanings of allocation information inserted in the second portion of a frame,
FIGS. 10 and 11 are tables showing sequences in which allocation information is stored for two different formats,
FIG. 12 is a block diagram of a receiver according to the invention,
FIG. 13 is a simplified block diagram of a transmitter which records the second digital signal on a magnetic record carrier,
FIG. 14 is a simplified block diagram of a receiver for producing a replica signal from the magnetic record carrier of FIG. 13,
FIGS. 15a-15d are diagrams of different arrangements of scale factors and samples in the third portion of a frame,
FIG. 16 is a block diagram of one preferred 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,
FIG. 20 is a table showing bit codings in an embodiment of the structure of FIG. 19,
FIG. 21 is a table showing a sequence for allocation information accommodated in a second frame portion associated with the first portion of FIG. 17, for a monaural mode,
FIGS. 22a-d are tables showing sequences for allocation information accommodated in a second frame portion associated with the first portion of FIG. 17, for a stereo intensity mode,
FIG. 23 is a diagram of a frame structure including an additional signal,
FIGS. 24 is a binary number diagram relating the sample with largest absolute value to an intermediate value used for scale factor computations,
FIG. 25 is a table showing quantization of scaled samples to form q-bit digital representations, and
FIG. 26 is a table showing dequantization of the q-bit digital representations.
DESCRIPTION OF THE PREFERRED EMBODIMENTS
FIG. 1 shows diagrammatically the second digital signal as generated by the transmitter and transmitted via the transmission medium. The second digital signal takes the form of the serial digital data stream. The second digital signal comprises
frames, two such frames, i.e. the frame j and the frame j+1, being given in FIG. 1a. The frames, such as the frame j, comprise a plurality of information packets IP1, IP2, IP3, . . . , see FIG. 1b. Each information packet, such as IP3, comprises N
bits b.sub.0, b.sub.1, b.sub.2, . . . , b.sub.N-1, see FIG. 1c.
Number of Packets
The number of information packets in a frame depends upon
(a) the bit rate BR with which the second digital signal is transmitted through the transmission medium,
(b) the number of bits N in an information packet, N being larger than 1,
(c) the sample frequency F.sub.s of the wide-band digital signal, and
(d) the number of samples n.sub.s of the wide-band digital signal.
The information which corresponds to these packets, and which after conversion in the transmitter belongs to the second digital signal, is included in one frame in the following manner. The parameter P is computed in conformity with the formula
##EQU1## If this computation yields an integer for P the number of information packets B in a frame will be equal to P. If the computation does not result in an integer some frames will comprise P' information packets and the other frames will comprise
P'+1 information packets. P' is the next lower integer following P. The number of frames comprising P' and P'+1 information packets is selected in such a way that the average frame rate is equal to F.sub.s /n.sub.s.
Hereinafter it is assumed that N=32 and n.sub.s =384. The table in FIG. 5 gives the number of information packets (slots) in one frame for these values for N and n.sub.s and for four values of the bit rate BR and three values for the sample
frequency F.sub.s. It is evident that for a sample frequency F.sub.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 frames comprise 35 information
packets (when BR is 128 kbit/s). This is also illustrated in FIG. 2.
FIG. 2 shows one frame. The frame comprises P' information packets IP1, IP2 . . . IP P'. Sometimes the frame comprises P'+1 information packets. This is achieved by 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 in the 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 sequence which comprise P'+1 information packets. By subtracting the numbers in the second and the third column from each other this yields the number of frames in the sequence comprising P' information packets. The (P'+1)th information packet
then need not contain any information, and may then comprise for example only zeroes.
It is obvious that the bit rate BR is not necessarily limited to the four values as given in the tables of FIGS. 5 and 6. Other (for example intermediate) values are also possible.
FIG. 2 shows that a frame comprises three frame portions FD1, FD2 and FD3 in this order. The first frame portion FD1 contains synchronising information and system information. The second frame portion FD2 contains allocation information. The
third frame portion FD3 contains samples and, when applicable, scale factors of the second digital signal. For a further explanation it is necessary to first describe the operation of the transmitter in the transmission system in accordance with the
invention.
The Transmission System
FIG. 4 shows diagrammatically the transmission system comprising a transmitter 1 having an input terminal 2 for receiving the wide-band digital signal S.sub.BB, which may be for example a digital audio signal. In the case of an audio signal,
this may be a mono signal or a stereo signal, in which case the digital signal comprises a first (left channel) and a second (right channel) signal component. It is assumed that the transmitter comprises a coder for subband coding of the wide-band
digital signal and that the receiver consequently comprises a subband decoder for recovering the wide-band digital signal.
The transmitter comprises an analysis filter 3 responsive to the digital wide-band signal S.sub.BB to divide the wide band into a plurality M of successive frequency subbands having band numbers m, where 1.ltoreq.m.ltoreq.M, which increase with
frequency. All these subbands may have the same bandwidth but, alternatively, the subbands may have different bandwidths. In that case the subbands may correspond, for example, to the bandwidths of the critical bands of the human ear. The analysis
filter generates subband signals S.sub.SB1 to S.sub.SBM, for the respective subbands. The transmitter further comprises circuits for sample-frequency reduction and block-by-block quantization of the respective subband signals, shown as the block 9 in
FIG. 4.
Such a subband coder is known and is described, for example, in the aforementioned publications by Krasner and by Theile et al. Reference is also made to the published European Patent Application 289,080, to which U.S. Pat. No. 4,896,362
corresponds.
For a further description of the operation of the subband coder reference is made to these publications, which are therefore incorporated herein by reference. Such a subband coder enables a significant data reduction to be achieved in the signal
which is transmitted to the receiver 5 through the transmission medium 4, for example a reduction from 16 bits per sample for the wide-band digital signal S.sub.BB to 4 bits per sample if n.sub.S is 384. This means that there are blocks of 384 samples
of the wide-band digital signal, each sample having a length of 16 bits. If a value M=32 is assumed, the wide-band digital signal is split into 32 subband signals in the analysis filter means 3. Now 32 (blocks of) subband signals appear on the 32
outputs of the analysis filter means, each block comprising 12 samples (the subbands have equal width) and each sample having a length of 16 bits. This means that on the outputs of the filter means 3 the information content is still equal to the
information content of the block of 384 samples of the signal S.sub.BB on the input 2.
The data reduction circuit 9 operates on the output of the filter 3 using the knowledge about masking. At least some of the samples in the 32 blocks of 12 samples, each block for one subband, are quantised more roughly and can thus be
represented by a smaller number of bits. In the case of a static bit allocation all the samples per subband per frame are expressed in a fixed number of bits. This number can be different for two or more subbands but it can also be equal for the
subbands, for example equal to 4 bits. In the case of dynamic bit allocation the number of bits selected for every subband may differ viewed in time, so that sometimes even a larger data reduction can be achieved, or a higher quality with the same bit
rate.
The subband signals quantised in the block 9 are applied to a generator unit 6. Starting from the quantised subband signals this unit 6 generates the second digital signal as illustrated in FIGS. 1 and 2. This second digital signal, as stated
hereinbefore, can be transmitted directly through the medium 4. Preferably, however, this second digital signal is first adapted in a signal converter (not shown), such as an 8-to-10 converter. Such an 8-to-10 converter is described in, for example,
European Patent Application 150,082 to which U.S. Pat. No. 4,620,311 corresponds. This converter converts 8-bit data words into 10-bit data words, and enables an interleaving process to be applied. De-interleaving, error correction and 10-to-8
conversion are then performed in the receiver.
Frame Format
The composition and content of the frames will now be explained in more detail. The first frame portion FD1 in FIG. 2 is shown in greater detail in FIG. 3. FIG. 3 shows that the first frame portion comprises exactly 32 bits and is therefore
exactly equal to one information packet, namely the first information packet IP1 of the frame. The first 16 bits of the information packet form the synchronising signal (or synchronising word), and may comprise for example only "ones" The bits 16 to 31
are system information. The bits 16 to 23 represent the number of information packets in a frame. This nu consequently corresponds to P', both for the frame comprising P' information packets and for frames comprising the additional information packet
IP P'+1. P' can be at the most 254(1111 1110 in bit notation) in order to avoid resemblance to the synchronising signal. The bits 24 to 31 provide frame format information.
FIG. 7 gives an example of the arrangement and significance of this information. Bit 24 indicates the type of frame. In the case of format A the second frame portion has another length (a different number of information packets) than in the
case of format B. As will become apparent hereinafter, the second frame portion FD2 in the A format comprises 8 information packets, namely the information packets IP2 to IP9 inclusive; and in the B format it comprises 4 information packets, namely the
information packets IP2 to IP5 inclusive. The bits 25 and 26 indicate whether copying of the information is allowed. The bits 27 to 31 indicate the function mode. This means:
a) the channel mode, which indicates the type of wide-band signal (as stated hereinbefore this may be a stereo audio signal, a mono audio signal, or an audio signal comprising two different signal components for example representing the same text
but in two different languages). FIG. 8 represents the channel mode. It illustrates how the signal components are divided between the two channels (channel I and channel II) in the aforementioned cases.
b) the sample frequency F.sub.s of the wide-band signal.
c) the emphasis which may be applied to the wide-band digital signal in the transmitter. The values 50 and 15 .mu.s are the time constants of the emphasis and CCITT J. The value 17 indicates a specific emphasis
standard as defined by the CCITT (Comite Consultative Internationale de Telegraphie et Telephonie).
The content of the frame portion FD2 in FIG. 2 will be described in more detail with reference to FIGS. 9, 10 and 11. In the A format the second frame portion contains eight information packets. This is based on the assumptions that the
wide-band digital signal S.sub.BB is converted into 32 subband signals (for every signal portion of the digital signal S.sub.BB), and that an allocation word having a length of four bits is assigned to every subband. This yields a total of 64 allocation
words having a length of 4 bits 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 number of subbands, so that now the second frame
portion comprises only 4 information packets.
FIG. 9 illustrates the significance of the four-bit allocation words AW. An allocation word associated with a specific subband specifies the number of bits by which the samples of the subband signal in the relevant subband are represented after
quantisation in the unit 9. For example, the allocation word AW which is 0100 indicates that the samples are represented by 5-bit words. Moreover, it follows from FIG. 9 that the allocation word 0000 indicates that no samples have been generated in the
relevant subband. This may happen, for example, if the subband signal in an adjacent subband has such a large amplitude that this signal fully masks the subband signal in the relevant subband. The allocation word 1111 is not used because it closely
resembles the sync word in the first information packet IP1.
FIG. 10 indicates the sequence, in the case that the frame mode is A, in which the allocation words AW, j,m associated with the two channels j, where j=I or II, and the 32 subbands of the sequence number m, m ranging from 1 to 32, are arranged in
the second frame portion. The allocation word AWl,1 belonging to the first subband signal component of the first and lowest subband (channel I, subband 1) is inserted first. After this the allocation word AWII,1 belonging to the second subband-signal
component of the first and lowest subband (channel II, subband 1) is inserted in the second frame portion FD2. Subsequently, the allocation word AWl,2 belonging to the first subband-signal component of the second and lowest but one subband (channel I,
subband 2) is inserted in the frame portion FD2. This is followed by the allocation word AW II,2 belonging to the second subband-signal component of the second subband (channel II, subband 2). This sequence continues until the allocation word AW II,4
belonging to the second subband-signal component of the fourth subband (channel II, subband 4) is inserted in the second frame portion FD2. The second information packet IP2 (slot 2) of the frame, which is the first information 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.
FIG. 10 merely gives the indices j-m of the inserted allocation word AW, j,m. FIG. 11 indicates the sequence for the allocation words in the case of a B-format frame. In this case only allocation words of the subbands 1 to 16 are inserted. The
sequence, as is illustrated in FIG. 10, corresponds to the sequence in which the separate samples belonging to a channel j and a subband m are applied to the synthesis filter means upon reception in the receiver. This will be explained in greater detail
hereinafter.
The serial data stream contains for example only frames in conformity with the A format. In the receiver the allocation information in each frame is then employed for correctly 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 conformity with the A format and frames in conformity with the B format. However, the frames in conformity with both formats may contain samples for all
channels and all subbands in the third frame portion. A frame in conformity with the B format then lacks in fact the allocation information required to derive the samples for the channels I or II of the subbands 17 to 32 from the third frame portion of
a B format frame.
The receiver comprises a memory in which the allocation information included in the second frame portion of an A format frame can be stored. If the next frame is a B format frame only the allocation information for the subbands 1 to 16 and the
channels I and II in the memory is replaced by the allocation information included in the second frame portion of the B format frame. The samples for the subbands 17 to 32 from the third frame portion of the B format frame are derived from the
allocation information for these subbands derived from the preceding A format frame and still present in the memory. The reason for the alternate use of A format frames and B format frames is that for some subbands the allocation information (in the
present case the allocation information for the higher subbands 17 to 32) does not change rapidly. Since during quantization the allocation information for the various subbands is available in the transmitter, this transmitter can decide to generate a B
format frame instead of an A format frame if the allocation information for the subbands 17 to 32 inclusive does not change (significantly). Moreover, this illustrates that now additional space 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 four information packets longer than the third frame portion of an A format frame. This enables the number of bits by which the samples in the lower subbands 1 to 16 are
represented to be increased, so that for these subbands a higher transmission accuracy can be achieved. Moreover, if it is required to quantize the lower subbands more accurately the transmitter can automatically opt for the generation of B format
frames. This may then be at the expense of the accuracy with which the higher subbands are quantized.
The third frame portion FD3 in FIG. 2 contains the samples of the quantised subband-signal components for the two channels. If the allocation word 0000 is not present in the frame portion FD2 for any of the subband channels this means that in
the present example twelve samples are inserted in the third frame portion FD3 for each of the 32 subbands and 2 channels. This means that there are 768 samples in total.
Scale Factors
In the transmitter the samples may be multiplied by a scale factor prior to their quantization. For each of the subbands and channels the amplitudes of the twelve samples are divided by the amplitude of that sample of the twelve samples which
has the largest amplitude. In that case a scale factor should be transmitted for every subband and every channel in order to enable the inverse operation to be performed upon the samples at the receiving end. For this purpose the third frame portion
then contains scale factors SF j,m, one for each of the quantised subband-signal components in the various subbands.
In the present example, scale factors are represented by 6-bit numbers, the most significant bit first, the values ranging from 000000 to 111110. The scale factors of the subbands to which these are allocated, i.e. whose allocation information
is non-zero, are accommodated in the leading part of the frame portion FD3 before the samples. This means that the scale factors are transmitted before the transmission of the samples begins. As a result rapid decoding in the receiver 5 can be achieved
without the necessity of storing all the samples in the receiver, as will become apparent hereinafter. A scale factor SF j,m can thus represent the value by which the samples of the signal in the j-th channel of the m-th subband have been multiplied.
Conversely, the number one divided by this value may be stored as the scale factor so that at the receiving end it is not necessary to divide the scale factors before the samples are scaled up to the correct values.
For the frame format A the maximum number of scale factors is 64. If the allocation word AW j,m for a specific channel j and a specific subband m has the value 0000, which means that for this channel and this subband no samples are present in
the frame portion FD3, it will not be necessary to include a scale factor for this channel and this subband. The number of scale factors is then smaller than 64. The sequence in which the scale 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 is therefore 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 I,5; SF II,5; SF II,6; . . . .
In this case the scale factors for the fourth subband of channel II and the sixth subband of channel I are not inserted. If the frame is a B format frame it may still be considered to insert scale factors in the third frame portion for all the
subbands and all the channels. However, this is not the only option. In this case it would also be possible to insert scale factors in the third frame portion of the frame for the subbands 1 to 16 only. In the receiver this requires a memory in which
all scale factors can be stored at the instant at which a previously arriving A format frame is received. Subsequently upon reception of the B format frame only the scale factors for the subbands 1 to 16 are replaced by the scale factors included in the
B format frame. The scale factors of the previously received A format frame for the subbands 17 to 32 are then used in order to restore the samples for these subbands included in the third frame portion of the B format frame to the correct scale.
The samples are inserted in the third frame portion FD3 in the same sequence as the allocation words and the scale factors, one sample for every subband of every channel in succession. According to this sequence, first all the first samples for
the quantised subband signals for all the subbands of both channels are inserted, then all the second samples, . . . etc. The binary representation of the samples is arbitrary, the binary word comprising only "ones" preferably not being used again.
The second digital signal generated by the transmitter 1 is subsequently applied to the transmission medium 4 by the output 7, and by means of the transmission medium 4 this signal is transferred to the receiver 5. Transmission through the
transmission medium 4 may be a wireless transmission, such as for example a radio transmission channel. Many other transmission media are also possible. In this respect optical transmission may be envisaged, for example over optical fibres or optical
record carriers, such as Compact-Disc-like media, or transmission by means of magnetic record carriers utilising RDAT or SDAT-like recording and reproducing technologies, 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, the receiver 5 comprises a decoder, which decodes the signal encoded in the coder 6 of the transmitter 1 and converts it into a replica of the wide-band digital signal supplied to the output 8. The essential information in
the incoming signal is contained in the scale factors and the samples. The remainder of the information in the second digital signal is merely required for a "correct bookkeeping", to allow a correct decoding. The receiver first derives the
synchronising and system information from the frames. The decoding process is then repeated for every incoming frame.
FIG. 12 shows a more detailed version of the receiver 5 of FIG. 4. The coded signal (the second digital signal) is applied to a unit 11 through the terminal 10. For every frame, the unit 19 first detects the sync words situated in the first 16
bits of the first frame portion. Since the sync words of successive frames are each time spaced apart by an integral multiple of P' or P'+1 information packets, the sync words can be detected very accurately. Once the receiver is in synchronism the
sync word can be detected in the unit 19. To accomplish this, a time window having, for example, a length of one information packet is opened after each occurrence of P' information packets, so that only that part of the incoming information is applied
to the sync word detector in the unit 19. If the sync word is not detected the time window remains open for the duration of another information packet because the preceding frame may be a frame comprising P'+1 information packets. From these sync words
a PLL in the unit 19 can derive a clock signal to control the central processing unit 18.
It is evident from the above that the receiver should know how many information packets are contained in one frame. For this purpose the switching means 15 are then in the upper position shown, to apply the system information to the processing
unit 18. The system information can now be stored in a memory 18a of the processing unit 18. The information relating to the number of information packets in a frame can be applied to the unit 19 over a control-signal line 20, to open the time window
at the correct instants for sync-word detection. When the system information is received the switch 15 is changed over to the lower position. The allocation information in the second frame portion of the frame can now be stored in the memory 18b.
If the allocation information in the incoming frame does not comprise an allocation word for all the subbands and channels this will have become apparent already from 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 the relevant information contained in the system information, the processing unit 18 will store the received allocation words at the correct location in the
allocation memory 18b.
It is obvious that in the present example the allocation memory 18b comprises 64 storage positions. If no scale factors are transmitted, the elements bearing the reference numerals 11, 12 and 17 may be dispensed with, and the content of the
third frame portion of a frame is applied directly by the connection 16 from the input 10 to a synthesis filter 21. The samples are applied to the filter 21 in the same sequence as the order in which the filter 21 processes the samples in order to
reconstruct the wide-band signal. The allocation information stored in the memory 18b is required in order to divide the serial data stream of samples into individual samples in the filter 21, each sample having the correct number of bits. For this
purpose the allocation information is applied to the filter 21 over the line 22.
The receiver further comprises a deemphasis unit 23 which subjects the reconstructed digital signal supplied by the filter 21 to deemphasis. For a | | |
