MXPA97005511A - Digital transmission system to transmit a digital audio signal that is in the form of samples of a specific length of word and that is presented at a specified speed of mues - Google Patents

Abstract

The present invention relates to a digital transmission system having a transmitter (11) and a receiver (12) for transmitting and receiving a digital audio signal. The digital audio signal is in the form of samples of a specific word length (WL) and is presented at a specific sample rate. The transmitter consists of a terminal (25, 30, 32) of input to receive the digital audio signal and to receive the first word (IW1) of information having a relation with the word specific length and a second word (IW2) of information that has a relationship with the specific speed of the sample. A formatting unit (28) is present for combining the digital audio signal and the first and second information words into a suitable serial data stream for transmission by means of a transmission medium (TRM, 12). The word length (WL) of the samples in the digital audio signal, expressed in number of bits, which is equal to n, where n is an integer greater than zero, and the sample rate is equal to 2P.Fs, where P is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, each group of frequency values includes 44.1 kHz and 48 kHz. The receiver consists of a sample rate converter (41) to convert the sample rate of the received signal.

Description

"DIGITAL TRANSMISSION SYSTEM TO TRANSMIT A SIGNAL AUDIO DIGITAL THAT IS IN THE FORM OF SAMPLES OF A SPECIFIC LENGTH OF WORD AND THAT IS PRESENTED TO A SPECIFIC SPEED OF SPECIMEN " DESCRIPTION OF THE INVENTION The invention relates to a digital transmission system having a transmitter and a receiver for transmitting and receiving a digital audio signal, the digital audio signal being in the form of samples of a specific length of speech and presents at a specific sample rate. The invention further relates to a transmitter for use in the transmission system, a receiver for use in the transmission system, and an information carrier obtained with the transmitter and a transmission signal. A transmission system as defined in the introduction paragraph is known from US-A 5,323,396, document DI in the list of related documents that can be found at the end of this description. The transmitter includes a subband encoder that divides a wideband digital audio signal into a plurality of subband signals, performs a quantization on the subband signals based on a psychoacoustic mask model for thus obtain a reduction of data in a quantity of audio information to be transmitted. Subsequently, quantized subband signals are combined into a composite transmission signal to allow transmission. Along with the reduced data audio information, a word of information related to the sample frequency of the wideband digital audio signal is transmitted in order to regenerate the wideband digital audio signal with the same sample frequency under the reception. The invention is directed to providing a transmission system as described in the introduction paragraph which is capable of processing and transmitting audio signals having a wide limit of sample frequencies and word lengths of the samples of the digital audio signal broad band. It is also another object of the invention to allow the reception of those audio signals transmitted, with the possibility of converting the sample frequencies of such audio signals to a desired sample frequency in the receiver. The transmission system according to the invention has a transmitter and a receiver for transmitting and receiving a digital audio signal, the digital audio signal is in the form of samples of a specific length of speech and is presented at a specific speed of sample, the transmitter consists of: - an input mechanism for receiving the digital audio signal and for receiving the first word of information having a relation with the word specific length and a second word of information having a relationship with the sample-specific rate, - a formatting mechanism for combining the digital audio signal and the first and second information words into a suitable serial data stream for transmission by means of a transmission medium, the word length of the samples in the digital audio signal, expressed in number of bits, which is equal to n, where n is an integer greater than zero, and the speed of stra is equal to 2P.Fs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz In addition, another object of the invention is to carry out the data compression step in the digital audio signal that is not based on a subband bandwidth data compression step. Preferably, the data compression step is a step without loss of data compression, such as variable length coding, Huffman coding, arithmetic coding or Lempel-ZiV coding. Because the data compression steps are much simpler to perform, compared to the known sub-band coding algorithm of the prior art. DESCRIPTION OF THE DRAWINGS These and other aspects of the invention will be apparent from and elucidated with reference to the embodiments described in the following description of the figure in which: Figure 1 shows the transmission system according to the invention; and Figure 2 shows another elaboration of part of the receiver in the system. DETAILED DESCRIPTION OF THE INVENTION Figure 1 shows a digital transmission system according to the invention, consisting of a transmitter 11 for transmitting a digital audio signal to t through a transmission medium TRM, as an information carrier 12 , to a receiver 13. An analog audio signal is supplied to an input terminal 1 of the transmitter 11, whose input terminal 1 is coupled to an input of a sigma-delta modulator 21. Under the influence of a very high sample frequency of N.Fs, where Fs is equal to 48 kHz or 44.1 kHz, or in exceptional cases 32 kHz, and where N can be chosen equal to 128, the 21 sigma-delta modulator converts the analog audio signal into samples with a limited word length varying from 6, as an example, to preferably 1 bit. In another description it will be assumed that the sigma-delta modulator 21 generates a 1-bit bitstream signal. Converting an analog audio signal into a bit stream of 1 bit has a number of advantages. Bitstream conversion is a high-quality coding method, with the possibility of high-quality decoding or low-quality decoding using a simpler decoding circuit. Reference is made with respect to the publications "A decimated digital filter for the analog-to-digital conversion of high-fidelity audio signals", written by J.J. van der Kam, document D2 in the list of related documents, and "A top-order topology for interpolar modulators for the oversampling of A / D converters", written by Kirk C.H. Chao et al, document D3 in the list of related documents. The bitstream signal is supplied to at least one low pass filter and the lower sampling unit. In the embodiment of Figure 1, an array of three low pass filter series and the lower sampling units 22, 23 and 24 are presented, all synchronized with the clock frequency L28.Fs and the derivatives thereof. The low pass filter and the lower sampling unit 22 consists of a low pass filter that filters the lowest 1/8 part of the frequency band of interest of the bitstream signal, which is 54. Fs Hz, in a band length of 8. Fs Hz and sample the bitstream signal by a factor of 8, so as to obtain an output signal with a sample rate of 16. Fs. The low pass filter and the lower sampling unit 23 in the same form, consist of a low pass filter that halves the frequency band of interest of the output signal of unit 22, which is now 8. Fs Hz at a bandwidth of 4. Fs Hz and sample this signal by a factor of 2, to obtain an output signal with half the sample rate of 8. Fs. The low pass filter and the lower sampling unit 24 in the same form consist of a low pass filter which halves the frequency band of interest of an output signal of unit 23, which is now 4. Fs Hz, at a bandwidth of 2. Fs Hz and sample this signal by a factor of 2, in order to obtain an output signal with half the sample rate of 4. Fs. In this way, at the output of the unit 24, the digital audio signal is available with a sample rate of 4. Fs and a word length, expressed in number of bits, equal to WL. This word length WL can have any value, depending on the accuracy of the calculation in units 22, 23 and 24. As an example, WL could be equal to 24. In this way, it should be noted that the digital audio signal presents in terminal 25 it has a word length and a sample rate, which depends on the choice made by the provider of the digital audio signal. The provider may prefer the 44.1 kHz value to be chosen for Fs. Other audio software providers may prefer 48 kHz to be chosen for Fs. further, the provider may choose more or less low pass filters and lower sample units to be used in order to derive the digital audio signal from a data stream present in the output of the converter 21. However, it should be understood, that according to the invention, the sample rate of the digital audio signal applied to the terminal 25 has a sample rate equaling 2p.Fs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz. In exceptional circumstances, Fs can be chosen equal to 32 kHz. The digital audio signal present in terminal 25 is supplied to an input 26 of a signal combination unit 28. The combination unit 28 further has the inputs 30, 32 and 34 to receive the first information word IW1, a second information word IW2 >; n and a synchronization word, respectively. The first information word is representative of a word length WL with which the samples of the digital audio signal are supplied to the input 26 and the second information word is representative of a sample rate, in the previous example 4. Fs of the digital audio signal supplied to the input 26. The synchronization words are supplied to the input 34 by means of a synchronization word generator 36. The combination unit 28 combines the information words IW1 and IW2 in the samples of the digital audio signal supplied to the input 26, in a composite signal. The synchronization words are added to the composite signal, and when necessary, an error correction coding step and a channel coding step are applied to the composite signal to obtain a transmission signal in the form of a current of serial data, suitable for transmission through a transmission medium TRM (disk 12). The combination of several input signals of the combining unit 28 into a composite signal can be performed by generating blocks of samples consisting of a synchronization word and a plurality of samples, wherein the information words are stored in each block of samples. samples The error correction coding step and the channel coding step can be carried out only in the samples, or in the samples that include the information words in a block. In addition, prior to the error correction coding and / or channel coding, a substantially no loss of data compression can be carried out on the samples of the digital audio signal applied to the input 26. The lossless encoders they have the advantage that they can compress data of the audio signal in such a way that, after the data expansion by a lossless decoder, the residual original bitstream signal can be reconstructed in a substantially lossless manner. This means that there is substantially no loss of information after compression-expansion. The lossless encoders can be in the form of a variable length encoder. Variable-length coders are well known within the technology. Examples of these variable-length coders are Huffman coders, arithmetic coders, and Lempel-Ziv coders. Reference is now made to this subject to the publications "A method for the construction of minimum redundancy codes", by D.A. Huffman, documented D5 from the list of related documents, "An introduction to arithmetic coding" by G.G. Langdon, D6 documents in the list of related documents, and "A universal algorithm for compression in data sequence" by J. Ziv et al, D7 documents in the list of related documents. The transmission means TRM may be an RF link, or a record carrier such as an optical disk or a magnetic record carrier, or even a solid state memory. Via the transmission means TRM, the transmission signal is supplied to the receiver 13. The receiver 13 consists of a detection unit 35 for the recovery of the transmission signal from the transmission medium TRM. The receiver L3 also consists of a sample rate converter 41, well known in the art. Reference is made to this matter in US-A 5,225,787, document D4 in the list of related documents.

The detection unit 35 is adapted to retrieve the first and second information words from the serial data stream and to retrieve the digital audio signal from the serial data stream using the first information word. As a result, samples of the digital audio signal having a word length WL are supplied to the output 38 with the sample rate substantially equal to 2p.Fs, which is in the present example 4. Fs. The second information word, representative of the sample frequency of the digital audio signal is supplied via line 42 to the sample rate converter 41 to control the conversion in the sample rate converter 41. The sample rate converter 41 is adapted to convert the sample rate of the samples to the digital audio signal supplied to its input 44 from the sample rate defined by the second information word IW2 at a second sample rate for obtain a digital audio signal converted into a speed of signal, which is supplied to an input 46. The second sample rate is equal to 2q.Fs ', where q is an integer greater than zero and Fs' is equal at a frequency value taken from a group of at least two frequency values, said group of frequency values include 44.1 kHz and 48 kHz. In exceptional cases, Fs' can be chosen equal to 32 kHz. Preferably, Fs ', is 48 kHz, as a sample rate conversion of a lower frequency Fs (e.g. equal to 44.1 kHz) at a higher frequency Fs' can be realized in a more simple way, with simpler filters. The word length WL 'of the samples supplied in the output 46 need not be the same as the word length WL of the samples of the received digital signal. Figure 2 shows an elaborated version of the sample rate converter 41. The converter 41 comprises a change register 51, first to first, a sample filter unit 53, and a variable retention circuit 55. In addition, a control signal generator 57 is present to generate a control signal at an output 59 for controlling the variable retention. The receiver consists of a frequency generator (not shown) that generates a frequency 128. Clock Fs, in response to the received information word IW2, to control the FIFO 51 sample filter 53. In addition, this frequency generator generates the frequency 2p.Fs which is supplied to an input 61 of the control signal generator 57, which has the form of a closed curve of digital control. The closed curve 57 consists of a frequency detector 63, a closed-loop filter 65 and a sigma-delta modulator 67. Samples of the digital audio signal recovered from the transmission signal by the detector 35 are supplied to the input of the FIFO 51. The FIFO 51 is required to count the variations in the speed with which the samples are recovered from the transmission medium. and are supplied to the detector 35 so as to obtain a sample data stream at the FIFO output 51 having a sample rate of 2? Fs, which is presently equal to 4. Fs. The sample filter 53 samples the digital audio signal, e.g. at a sample rate of 64 Fs. The word length WL 'of the samples at the output of the filter 53 may be greater than the word length WL. The already sampled digital audio signal is supplied at the input of the variable retention circuit 55, which supplies the output samples at a rate of 2q.Fs', in response to the control signal applied to the signal input 70 -trol. In response to the control signal, it will be decided whether the next output sample is obtained by taking a new input sample or by repeating the previous output sample. The variable retention circuit 55 operates under the influence of a clock signal, e.g. 128. Fs', generated internally by the receiver. The conversion process in the variable retention circuit 55 is controlled by the sigma-delta modulator 67 in the closed control curve 57, which is synchronized by a crystal oscillator 69 having an oscillation frequency Fx. The signal output of the sigma-delta modulator 67 is in the form of '+1' and '-1' pulses and has a modulated pulse density version of its input signal. If for example, the input signal has a DC value of 0.5, then the sigma-delta modulator 67 will generate three pulses '+1' and a pulse '-1', so that in the average. { 3. (+ L) + 1. (-1)} /4=0.5. If the clock frequency of the sigma-delta modulator 67 is Fx, it will generate pulses in one second. Suppose now that this clock frequency is chosen to be the same as the sample output frequency of the sample rate converter, then the output pulses of the sigma-delta modulator 67 can be used to control the conversion process. The signal input of the sigma-delta modulator 67 into a DC value that is dependent on the sample input and output frequencies of the sample rate converter. The variable retention circuit 55 is controlled in such a manner by means of the pulses supplied by the sigma-delta modulator 67 that a pulse '+1' means that the previous output sample is repeated and a pulse '-1' means that He took a new sample of input. Each second must convert samples 64. Fs input to samples 2q.Fs output (2q will generally be chosen equal to 64 also, in the present example). When the input sample frequency of the sample rate converter is less than the output sample frequency of the converter, all the input samples of the feedback circuit will be used as its inputs to generate an output signal of the circuit 55 retention. This means that the 67 sigma-delta modulator has to generate pulses 64. Fs '-1' in one second. The remaining 2q.Fs'64.Fs samples of output are obtained by repeating some of their input samples, as by retaining the previous output sample. Therefore, the 67 sigma-delta modulator has to generate pulses 2q.Fs'-64.Fs '+ l' in one second. In addition, the receiver may consist of a sound profiler 72, followed by the DA converter 74 and the low pass filter 76. The sound profiler 72 and the DA converter 74 also operate under the influence of the frequency 128. Fs'. The sound profiler converts the digital signal applied to its input into a 1-bit bitstream signal having a sample frequency of 64. Fs', which is subsequently DA converted to converter 74 and filtered on filter 76 with in order to obtain an analog audio signal at output 80 with a bandwidth of 20kHz, as an example. The transmission system as described above has the advantage that the audio signals of various word lengths WL and sample rates, related to Fs can be transmitted through the transmission medium, with the possibility of receiving those signals and convert them to a set frequency, related to Fs'. While the invention has been described with reference to the preferred embodiments thereof, it should be understood that these are not limiting examples. Therefore, several modifications may be apparent to those enabled in the technology, without departing from the invention panorama, as defined in the clauses. In addition, the invention lies in each of the novel features or in the combination of features. RELATED DOCUMENTS (DI) US-A 5,323,396 (PHN 13,241) (D2) "A decimated digital filter for the analog-to-digital conversion of high-fidelity audio signals", written by J.J.van der Kam, at Philips Techn. Rev. 42, no 6/7, April 1986, pp. 230-8. (D3) "A high-order topology for interpolar modulators for the oversampling of A / D converters", written by Kirk C.H. Chao et al in IEEE Trans. in Circuits and Systems, Vol 37, no. 3, March 1990, pp. 309-18 (D4) US-A 5,225,787 (PHN 13.677) (D5) "A method for building minimum redundancy codes" by D.A. Huffman in Proc. of the IRÉ, Vol. 40 (10), September 1952 (D6) "An introduction to arithmetic coding" by GGLangdom, IBM J. Res. Development, Vol. 28 (2), March 1984. (D7) "A universal algorithm for the compression in data sequence" by J. Ziv et al, IEEE Trans. in Inform. Theory, Vol. IT-23, 1977. 15 20 or NOVELTY OF THE INVENTION Having described the invention, it is considered as a novelty, and therefore, the content of the following clauses is claimed as property. CLAUSES 1. A digital transmission system having a transmitter and a receiver to transmit and receive a digital audio signal, the digital audio signal is in the form of samples of a specific length of speech and is presented at a specific speed of word, the transmitter consists of :: - an input mechanism for receiving the digital audio signal and for receiving the first word of information having a relation with the word specific length and a second word of information having a relationship with the sample-specific rate, - a formatting mechanism for combining the digital audio signal and the first and second information words into a suitable serial data stream for transmission by means of a transmission medium, the word length of the samples in the digital audio signal, expressed in number of bits, which is equal to n, where n is an integer greater than zero, and the sample rate ra is i? jual to 2P.Fs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz. 2. A digital transmission system as claimed in clause 1, where Fs is equal to a frequency value taken from a group of at least three frequency values, said group of frequency values includes 32 kHz, 44.1 kHz and 48 kHz 3. A digital transmission system as claimed in clause 1 or 2, wherein the formatting mechanism consists of a channel coding mechanism for channel coding of at least the digital audio signal to thereby ob-have a stream of data in series. 4. A digital transmission system 'as claimed in clause 1 or 2, wherein the formatting mechanism consists of an error correction coding mechanism for the error correction coding of at least the digital audio signal to obtain a stream of data in series. 5. A digital transmission system as claimed in clause 1, 2, 3 or 4, wherein the transmitter further includes a sigma-delta modulator and at least one low pass filter and a lower sample unit, the modulator sigma-delta is adapted to receive an analog audio signal and to supply an audio signal of a bitstream of 1 bt in response to the same, the at least one pass filter ba and the lower sample unit are adapted to sample the signal of the bit stream of 1 bit so as to obtain a

Claims (1)

1. Digital audio signal already sampled, and to supply the digital audio signal to the input mechanism. 6. A digital transmission system as claimed in clause 5, wherein the low pass filter and the lower sample unit sample with a factor 2r, where r is an integer greater than zero. 7. A digital transmission system as claimed in any of the preceding clauses wherein the receiver comprises: - an input mechanism for receiving the stream of data in series from the transmission medium; - a recovery mechanism for recovering the first and second information words from the serial data stream, for recovering the digital audio signal from the serial data stream using the first information word; - a sample rate conversion mechanism for converting the sample rate of the samples to the digital audio signal supplied by the retrieval mechanism from the sample rate defined by the second information word at a second sample rate in order to obtain a digital audio signal converted into a sample rate, where the second sample rate is equal to 2q.Fs, where q is an integer greater than zero and Fs' is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz, the samples in said digital audio signal converted into a sample rate have a specific word length, and - a output mechanism for supplying the digital audio signal converted to a sample rate at said second sample rate. 8. The digital transmission system as claimed in clause 7, where Fs' equals 48 kHz. 9. The digital transmission system as claimed in clause 7 or 8, wherein the receiver also consists of a D / A conversion mechanism to convert the digital audio signal converted into a sample rate into an analog audio signal . 10. The digital transmission system as claimed in clause 7, 8 or 9, wherein the sample rate conversion mechanism consists of: - a sample mechanism for sampling the digital audio signal to obtain a digital signal raues-processed audio that has a sample rate equal to 2q.Fs, - a variable retention mechanism to variablely retain the samples of the digital audio signal routed to obtain the digital audio signal converted into a digital audio signal. Sample speed. 11. A digital transmission system as claimed in any of clauses 7 to 10, wherein the input mechanism consists of a channel decoding mechanism for the channel decoding of the serial data stream. 12. A digital transmission system as claimed in any of clauses 7 to 10, wherein the input mechanism consists of an error correction mechanism for performing an error correction step in a signal applied to the input of the error correction mechanism. 13. A transmitter for use in a transmission system as claimed in any of clauses 1 to 6, characterized by those characteristics in clauses 1, 2, 3, 4, 5 or 6 that characterize the transmitter. 14. A transmitter as claimed in clause 13, wherein the transmitter is in the form of a recording apparatus for recording the serial data stream in a track on a record carrier, the formatting mechanism further comprising mechanisms of writing to write the serial data stream in said track in the record carrier. 15. A receiver for use in a transmission system as claimed in any of clauses 7 to 12, characterized by those characteristics in clauses 7, 8, 9, 10, 11, or 12 that characterize the receiver. 16. The receiver as claimed in clause 15, wherein the receiver is in the form of a reproduction apparatus for reproducing the data stream from the track in the record carrier, the input mechanism further comprises reading mechanisms for reading the serial data stream from said track on the record carrier. 17. The record carrier obtained with the transmitter as claimed in clause 14, a serial data stream that is registered in a track in said record carrier, said serial data stream consists of samples of a digital signal of audio and a first and second information words included in said stream of serial data, the first information word has a relation to the word length of the samples in said digital audio signal and the second information word has a relationship with the sampling rate of the samples in said digital audio signal, the word length of the samples in the digital audio signal, expressed in number of bits, is equal to n, where n is an integer greater than zero, and the sample rate of the samples in the digital audio signal is equal to 2pFs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two values s frequency, said group of frequency values includes 44.1 kHz and 48 kHz. 18. A method for transmitting a digital audio signal, the digital audio signal is in the form of samples of a specific length of speech and is presented at a specific sample rate, the transmission method consists of the steps of : - receiving the audio signal, - receiving a first word of information that is related to the word-specific length and a second word of information that is related to the specific sample rate, - combining the digital audio signal and the first and second information words in an appropriate data stream for transmission through a transmission medium, the word length of the samples in the digital audio signal, expressed in number of bits, is equal to n, where n is an integer greater than zero, and the sample rate is equal to 2pFs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz. 1 . If a signal is found in a digital audio signal and a first and second information words, the first word of information has a relation to the word length of the samples in said signal. digital audio and the second word of information has a relationship with the sample rate of the samples in said digital audio signal, the word length of the samples in the digital audio signal, expressed in number of bits, is equal to , where n is an integer greater than zero, and the sample rate of the samples in the digital audio signal is equal to 2pFs, where p is an integer greater than zero and Fs is equal to a frequency value taken from a group of at least two frequency values, said group of frequency values includes 44.1 kHz and 48 kHz. 20. A transmission system as claimed in clause 7, wherein the transmitter further comprises a lossless compression mechanism for performing a data compression step substantially without loss in the digital audio signal for obtaining a digital audio signal of compressed data for transmission through a transmission medium, the receiver further comprises a lossless expansion mechanism for performing a data expansion step in the digital audio signal of compressed data in the transmission signal to obtain a replica of the digital audio signal.