CA1179794A - Apparatus and method of reducing the bit rate of pcm speech - Google Patents

Abstract

ABSTRACT OF THE DISCLOSURE

The present invention relates to an apparatus and method of reducing the bit rate of PCM speech in a multiplexed voice-channel system; for each of the M
time-division-multiplexed channels it reduces the bit rate from the (original) 8 bits/sample PCM represen-tation into a f bits (per sample representation where f = 6,5,4 or 3); at the receiving end the device is used in its rate expansion function to retrieve the PCM repre-sentation with 8 bits per sample.

Description

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FIELD OF THE INVENTION
The present invention relates generally to PCM
(Pulse Code Modulation) telecommunications and, more particularly, to a method and apparatus for reducing the bit rate of speech channels already encoded in PCM.
BACKGROUND OF THE INVENTION
PCM is a well-known method of transmitting telephone speech which consists of periodically sampling the amplitude of voice-frequency signal and translating these amplitudes into digital form. This method is well suited to the transmission of several voice channels using time division multiplex.
PCM transmission of telephone spee~h is exten-sively used in North America and in the world~ Several schemes have been proposed for the purpose of reducing the bit rate of PCM speech: in particular, the tech-nique known as NIC (Nearly Instantenous Companding, invented by Deutweiller et Messerschmitt and described in U.S. patent No. 3,945,002 issued March 16, 1976) which reduces from 8 bits per sample to 6 bits per sample with the provision that three overhead bits be sent every N samples (where N is some fixed integer between 6 and 128).
OBJECTS OF THE INVENTION
An object of the invention is to reduce the rate of the PCM speech representation from an 8 to an f ~it/
sample while preserving a high subjective quality, in ,~1 ~7~

particular, a better quality than that which would be obtained by simply dropping the 8-f least significant bits of the PCM representation.
It is a further object of the present in-vention to provide a reduced format which can be pre-scribed dynamically that is a different format can be prescribed from sample to sample in a given voice channel. In addition in a time-division-multiplex embodiment a different format can be prescribed from channel to channel.
It is still a further object of the present invention to reduce the bit rate of PCM speech in a multiplexed voice-channel system to a digital speech using 6,5,4 or 3 bits per sample format.
For each of the time-division-multiplexed channels, say channel m (where m = 1,2,3...M), the input PCM sample is denoted at time n by Xn and Sn which are respectively the sample magnitude (i.e.: absolute value:
Xn = 0,1,2,...,127) and its sign (Sn = 0,1 where 0 stands for a negative sample and 1 for a positive one).
After the rate reduction operation, the sample is transmitted under a new representation namely: Zn and Sn which are respectively the "tag" (some number Z = 0,1,2,...,2f 1-1) and the sign.
Finally at the receiving end, a PCM word is retrieved denoted Xn and Sn which are respectively the (output) sample magnitude and its sign.

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While the technique does not interfere with the transmission of the sign Sn, the retrieved magnitude Xn might be slightly different from the input magnitude Xn but without resulting in subjective degradation.
The above objects and oth~rs are achieved with the present invention where in a pulse code modulation (PCM) transmission system for communicating digitized message samples between a transmitting station and a receiving station, the transmitting station includes an apparatus for reducing the bit rate of PCM samples from a 8-bit per sample format to a f-bit per sample format;
the apparatus comprises a) input means for receiving and breaking down each input PCM sample received to a magnitude Xn, wherein Xn is the absolute value of the sample at time n, and to a sign Sn, wherein Sn is the sign of the sample at time n;
b) feedback means for transmitting a retriev-able sample X 1' wherein Xn is the magnitude of a 0 retrieved sample at time n;
c) means for extracting an envelope infor-mation from Xn;
d) means for extracting information relating to a sign change between consecutive samples to provide 5 a zero-crossing mode Mn;
e) means for computing the difference Dn at time n between Xn and Xn l;

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f) selecting means for transmitting - X when X is in the vicini.ty of a sign change or for transmitting Dn otherwise;
and - a quantizer value reflecting a zero-crossing mode Mn and a signal energy indicated by said envelope extracting . means;
g) first mapping means for establishing a correspondence between Xn or Dn transmitted and a binary word having f-l number of bits;
h) second mapping means for providing said retrieved sample Xn or a retrieved difference D from information obtained from the binary word, the format, the zero-crossing mode Mn and the envelope extracting means; and i) means combining Sn and Xn to provide an output PCM sample.
The present invention also relates to a method for reducing the bit rate of digitized message PCM
samples.
The invention will be more thoroughly under-stood after reading the following detailed description in conjunction with the drawings wherein:
E'igure 1 is a simplified block diagram of the system according to the present invention;

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Figure 2 is a block diagram of the device according to the present invention;
Figure 3 is a graph giving an envelope estimate;
Figures 4, 5 and 6 give probability distri-bution of PCM magnitudes and reencoder mappings for the cases J = 0,1,2,.~.,11.
In PCM, the speech is band-pass filtered, between 300 and 3400 Hz and sampled at 8000 samples per second. Each sample is then rounded off by a nonuniform quantizer and coded in binary by an 8-bit word.
Considering the sample at time n, the most significant bit is the sign bit, denoted S . The other seven bits represent the magnitude and is denoted Xn.
In decimal, this magnitude can be expressed as an integer between 0 and 127.
Sn - (negative), 1 (positive) Xn = ~ 1, 2, ..., 127 In the 255 companding law the magnitude Xn is essential-ly the log of the actual voltage of the sample. Thisarrangement provides a wide dynamic range for the speech signal and a signal to noise ratio which is basically constant.
The present invention relates to a rnethod of reducing the bit rate of PCM speech in a multiplexed voice-channel system.
Figure 1 illustrates the overall use of the device which is composed of two parts: the RR module 10 (Rate Reduction) and the RE module 12 (Rate Expansion).
While the Rate Expansion function which takes place at the receiving end requires only the RE module, the Rate Reduc-tion at the transmitting end requires that both the RR and RE modules be operative as illustrated on Figure

2. In this case, the RE module is used to replicate locally the expansion process which takes place at the receiving end, thereby providing a Eeedback to the RR
module. This feedback consists of two quantities:
First, Xn 1 which is a replicate of the retrieved magni-tude at time n-l and, second, En 1 which is an estimate of the signal's envelope also at time n-l.
The RR module 10 includes a preprocessing module 14 in which the input PCM sample at time n is broken down into its magnitude, Xn, and its sign, Sn.
The order of the binary bits of Xn are further reversed, (least significant bit leading) for subsequent serial arithmetic processing. Finally the magnitude is delayed during one sampling time to allow the sign Sn+l of the forthcoming samp]e to reach the ZCM (zero-crossing-mode) module 16.
This module 16 outputs a flag Mn (M = 0,1).
Mn = 1 signals the fact that Xn is in the vicinity of a zero crossing of the speech waveform (i.e.: a sign change between two consecutive samples). More pre-cisely: Mn = whenever Sn_2 = Sn_l = Sn Sn+l otherwise Mn = 1.
Prior to entering the processing module 18, the difference D = Xn-Xn l is computed. One ch~racter-istic feature of the procedure is that Xn will be pro-cessed and -transmitted whenever Xn is in the vicinity of a zero crossing (Mn = l). This is the direct scheme.
On the other hand, when Xn is far from any zero crossing (i.e.: usually at waveform maxima, Mn = )~ it is the quantity Dn which is transmitted following adequate pro-cessing. This is the differential scheme. This featureof the procedure takes advantage of properties of the PCM logarithmic companding law.
The processing module 18 selects therefore either Xn or D on the basis of M . The object of this module is to provide two outputs: first, a properly scaled and bounded Xn or Dn (depending on which has been selected) and second an index, (j = 0,1,2,...,J-l), which reflects both the zero crossing mode Mn and the signal energy as indicated by the envelope information En l Basically, for a given Mn a large value of J
indicates a large input signal and a small value a small signal. Typically J = 0,...,7 when Mn = 1 and J = 8,9, ...,ll when Mn =
The purpose of the mapping module 20 is to provide a correspondence between a given magnitude X
(or difference Dn) and a tag Zn with a prescribed number of bits. This number of bitsis f-l where f is the format specified by the user at time n. The mapping module is typically implemented using a Read-Only--Memory (ROM). For each for~lat f, there are J
different possible mapping between Xn (or Dn) and the tays. A preferred embodiment consists of using Xn (or Dn) with j as the address field of the ROM. In this case, the ROM is composed of J partitions each containing one of the J possible correspondences or mappings between Xn (or Dn) and the tags Z .
The low bit rate output of the device is a f bit word made of Sn+l and the tag Zn with f-1 bits.
The low bit rate output of the device is an f bit word with the following breakdown: 1 bit for the sign Sn+l and f-l bits for the tag Zn To simplify timings in the preferred embodiment of the device the output bit stream has the same data rate as the input bit stream~
but of the 8 output bits corresponding to a particular sample the information has now been condensed within the f leading bit positions while the remaining 8-f positions are dummy bits. The device can be used in several ways to achieve bit rate reduction. The simplest way is to use the same (fixed) format for any channel and any sample. In this case, one has simply to delete the dummy bits to achieve a bit rate reduction according to the ratio f/8. A more efficient use of the device is possible in the context of a ~:~7~7~ ~
- 8.a -Digital Speech Interpolation (DSI) system. A DSI
system would use a set of speech detectors to keep track of the speech activity in each channel and determine an optimal format f for each channel. The DSI system would prescribe to the device of this invention the format f for a given channel and collect at the binary output of the said device the f information bits (leading bits) for that channel. The DSI system would then recombine the information bits of the active channels to achieve a greater overall bit rate reduction ratio. The low bit rate word which proceeds either from a distant transmitter or from the local RR function is interpreted by the reverse mapping module 22. This module is also well suited for a Read-Only-Memory type of implementation. The address field of this seecond memory hereafter referred to as ROM2 is made up of Zn~ of the format f, the zero~crossing mode Mn, and the envelope estimate En 1 The content of the memory corresponding to a particular address provides directly the retrieved magnitude ~n' if Mn = 1, or the retrieved difference Dn if M = 0.
The restoration module 24 retrieves ~n from the previous magnitudes ~n 1 and the transmitted difference ~n whenever the transmission is in the differential mode.

~n ~n-l Dn The envelope extraction module 26 receives the retrieved magnitude Xn and extracts an envelope infor-mation according to -the following recursive procedure:
En = max ~n-l 1, n ~

The past processing module 28 combines S and X to obtain the output PCM sample. In particular the order of the bits of the magnitude are reversed (most significant bit leading) and properly complemented to yield the standard PCM representation.
Other recursive forms may be selected for this estimate such as low order filtering of Xn 1 This form is selected, however, for both its simplicity of implementation and its ability to swiftly increase upon voicing attacks. Advantageously, this estimate does not have any time constant in terms of increase. By contrast, the rate of decrease is slowed down.
Figure 3 exemplifies this behaviour on an actual sequence of PCM magnitudes. Three pitch periods are represented. It can be seen that, with this par-ticular time constant, the estimate is able to maintainreasonably well the pick value over the pitch interval.
Some of the implemen-tation simplicity of this estima-te would be lost if a slower time constant was sought for.
Let us now turn to the question of using this envelope information to control the quantization stage.
The approach is that of using a set of quantizers and to switch on one or the other according to the value of ~L~L7~7~'~

the envelope. It would be both prohibitive and useless to use 128 quantizers, one for each value of the en-velope E . A set of 8 quantizers is retained and labelled from O to 7. The rule for selecting the proper quantizer is the foLlowing. Quantizer k is to be used whenever the three most significant bits of En corre-sponds to k in decimal, or equivalently, when the inteyer part of En/16 is equal to k. This arrangement has the merit to rela-te the quantizer index to the chord numbers of the PCM companding law. Hence, it may be said that quantizer k will be used whenever the sample-magnitude, Xn, to be quantized is most likely to occur in the kth (or a possibly lower) chord.
Graphical representation of the reencoders are given in Figures 4, 5, 6 with the probability distri-butions for magnitudes and increments conditioned upon the various values of k.

X = 0,127 TABLE FOR J = 1 ..____ _ _ . .__ _ . ~ =_ -~ ~ __ _=__ _ _ _ Xn = Zn = O* . O* O*
1 0* O l* l*
2 O 1* 2* 2*

3 1 2* 3* 3*

4 1* 2 4* 4*
6 1 -3 6** 6-~
7 . 2 4 8~ ~37*
9 2 4 8 ,~*
2 ~ 4 .~9* 10*
11 - 2* 5 9 11*
12 2 . 5* 10* - . 12*
13 2 5 10 13*
14 2 5 10 14*
- 2 6 11 15*
16 3 6 1 1* 1 6*
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1 27 . .. . .. .. ~ . 1 5 .

Table 1 This table illustrates the way the content of ROMl and ROM2 is extracted from the graphs of Figs. 4,

5, 6. The above example relates to the first case, J = 0, namely the transcoding of Xn into Zn by ROMl and the reverse transcoding of Zn back into Xn by ROM2.
Suppose for ins-tance Xn = 7 (and J = 0) than ROMl will provide the outputs Zn = 1,4,7,7, corresponding re-spectively to formats 3,4,5,6. If the format was, say f = 4, the ROM2 would receive Zn = 4 (and f = 4; J = 0) it would provide the output Xn = 8, that is, the value of X in the table corresponding to the value Zn = 4 that is noted by an asterisk. For convenience, the values Xn (or Dn) corresponding to a Zn with an asterisk (i.e.:
the prototypes) have been listed in their natural order in Tables 2, 3 and 4 for every State J = 1,2,... ,15 and every format f = 3,4,5 and 6. Note that for format f there are 2f l such pro-totypes.

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f: Prototypes for j = 0 (Xn, k=0) P() = .084

6 0 1 2 3 4 5 6 7 ___________________________________________________________ f: Prototypes for j = 1 (Xn, k=l) P(l) = .092 ___________________________________________________________ f: Prototypes for j = 2 (Xn, k=2) P(2) = .074 4 5 17 24 32 37 44 51 . 61 ___________________________________________________________ f: Prototypes for j = 3 (Xn, k=3) P(3) = .062 4 13 29 38 47 53 61 69 a4 .________ __________. _______ _____________________________ Table 2 i'7~

f: Prototypes for j = 4 (Xn, k=4) P(4) = .060 66 67~ 69 7274 77 80 81 ___________________________________________________________ f: Prototypes for j = 5 (Xn, k=5) P(5) = .071 ___________________________________________________________ f: Prototypes for j = 6 (Xn~ k=6) P(6~ = .048 ___________________________________________________________ f: Prototypes for j = 7 (Xn, k=7? P(7) = .008 3 64 9~ 104 118 ____________________________________ ______________________ Table 3

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f: Prototypes for j = 8 (Dn, k=0) P(8) = .071 ~ -6 -1 4 11 ___________________________________________________________ f: Prototypes for j - 9 (Dn, k=l) p(9) = .06B

___________________________________________________________ f: Prototypes for j = 10 (Dn, k=2,3~4,5,6) P(10) = .344 - 4-17 -5 2 11 23 39 57 . 78 2 . 7 13 22 34 47 60 78 -7 -6 -5 -4 ~ -3 -2 -1 0 ___________________________________________________________ f: Prototypes for j = 11 (Dn, k=7) P(ll) = .014 12 17 25 . 36 46 56 75 95 ____________________________________ ______________________ Table 4

Claims (13)

The embodiments of the invention in which an exclusive property or privilege is claimed are defined as follows:

1. In a pulse code modulation (PCM) trans-mission system for communicating digitized message samples between a transmitting station and a receiving station, said transmitting station including an appa-ratus for reducing the bit rate of PCM samples from a 8-bit per sample format to a f-bit per sample format, said apparatus comprising a) input means for receiving and breaking down each input PCM sample received to a magnitude Xn, wherein Xn is the absolute value of the sample at time n, and to a sign Sn, wherein Sn is the sign of the sample at time n;
b) feedback means for transmitting a retriev-able sample ?n-1, wherein ?n is the magnitude of a retrieved sample at time n;
c) means for extracting an envelope infor-mation from ?n;
d) means for extracting information relating to a sign change between consecutive samples to provide a zero-crossing mode Mn;
e) means for computing the difference Dn at time n between Xn and ?n-1;
f) selecting means for transmitting - Xn when Xn is in the vicinity of a sign change or for transmitting Dn otherwise;
and - a quantizer value reflecting a zero-crossing mode Mn and a signal energy indicated by said envelope extracting means;
g) first mapping means for establishing a correspondence between Xn or Dn transmitted and a binary word having f-1 number of bits;
h) second mapping means for providing said retrieved sample ?n or a retrieved difference ? from information obtained from said binary word, said format, said zero-crossing mode Mn and said envelope extracting means; and i) means combining Sn and ?n to provide an output PCM sample.

2. In a PCM system as defined in Claim 1 further comprising restoration means connected to said second mapping means for retrieving ?n by adding Dn to the retrieved magnitude ?n-1

3. In a PCM system as defined in Claim 1, wherein said first mapping means is composed of a Read-Only-Memory.

4. In a PCM system as defined in Claim 1, wherein said second mapping means is composed of a Read-Only-Memory.

5. In a PCM system as defined in Claim 1, said receiving station including a rate expansion appa-ratus consisting of means defined under c), h) and i).

6. In a PCM system as defined in Claim 5, said rate expansion apparatus further including resto-ration means connected to said second mapping means for retrieving ?n by adding Dn to the last retrieved magni-tude ?n-1.

7. In a PCM system as defined in Claim 5, wherein said second mapping means of said rate ex-pansion apparatus is composed of a Read-Only-Memory.

8. A method for reducing the bit rate of digitized message PCM samples between a transmitting station and a receiving station of a time-division-multiplexed voice-channel system for a 8-bit per sample format to a f-bit per sample format comprising the steps of:
a) receiving and breaking each input PCM
sample to a magnitude Xn and a sign Sn, wherein Xn is the absolute value of the sample at time n and Sn is the sign of the sample at time n;

b) transmitting a retrievable sample ?n-1, wherein ?n is the magnitude of a retrieved sample at time n;
c) extracting an envelope information from ?n;

d) extracting information relating to a sign change between consecutive samples to provide a zero-crossing mode Mn;
e) computing the difference Dn at time n between Xn and ?n-l;
f) transmitting - Xn when Xn is in the vicinity of a sign change or for transmitting Dn otherwise;
and - a quantizer value reflecting a zero-crossing mode Mn and a signal energy indicated by the envelope extracting step;

g) establishing a correspondence between Xn or Dn transmitted and a binary word having f-1 number of bits;
h) providing said retrieved sample ?n or difference ? from information obtained from the binary word, the format, the zero-crossing mode Mn and the envelope;
i) combining Sn and ?n to provide an output PCM sample.

9. In a PCM system as claimed in claim 1, wherein said selecting means receives said signal Mn from said means for extracting information and trans-mits one of said Xn and Dn in response to the level of said signal Mn.

10. In a PCM system as claimed in claim 9, wherein said signal Mn is at a high level when said Xn is smaller than a predetermined amount and said signal Mn is at a low level when said Xn is larger than said predetermined amount.

11. In a PCM system as claimed in claim 10, wherein the level of said signal Mn is determined to be low whenever said sign of the sample meets the following relationship, Sn-2=Sn-1 =Sn=Sn+1, wherein Sn-2 is the sign of the sample at time n-2, Sn-1 is the sign of the sample at time n-1, and Sn+1 is the sign of the sample at time n+1.

12. In a PCM system as claimed in claim 1, wherein said first mapping means establishes said correspondence between Xn or Dn transmitted and said binary word having f-1 number of bits in response to said envelope information from Xn.

13. In a PCM system as claimed in claim 12, wherein said selecting means receives said envelope information from said means for extracting information and outputs an index signal to said first mapping means, said index signal being generated in accordance with said signal Mn and said envelope information.