Apparatus for time-scaling in communication products

A communication system using voice compression includes a transmitter base station (113) and a selective call receiver (112). The transmitter base station (113) includes an input device (204) to receive an audio voice message which is stored in a memory (209). A processing device (208) digitizes the audio voice message to provide an input signal which is divided into a sequence of equivalent time frames, and differences in a short-term frequency spectrum between adjacent time frames is determined to derive distance measurements. A speed factor is computed as an average of the distance measurements, and the input signal is time-scales in accordance with the speed factor and then transmitted by a transmitter (102). The selective call receiver (112) includes a receiver (105) which receives the time-scaled signal, a processing device (115, 106) which demodulates and expands the time-scaled signal in accordance with the speed factor to provide a reconstructed signal which is amplified by an amplifier (108) into an audio signal.

TECHNICAL FIELD 
This invention relates generally to time scaling techniques, and more 
particularly to a method and apparatus for improved time scaling utilizing 
a speed factor derived from an input signal. 
BACKGROUND 
Time-scaling methods are used to compress and expand input signals, in 
particular, speech input signals. Further, time-scaling techniques when 
used with communication systems allow for a more bandwidth efficient 
system than other techniques. Time-scaling of voice signals also generally 
presents tradeoffs between voice compression ratios and speech quality 
upon reconstruction. Existing time-scaling techniques have failed to fully 
consider that different speakers talk at different rates or speeds. Thus, 
using existing techniques, speech of faster speakers might be compressed 
too much and thus provide lower quality reconstructed speech and speech of 
slower speakers would not be compressed enough, thus causing inefficient 
compression. Also, certain applications like dictation and sound editing 
require that all messages be played at the same speed irrespective of how 
fast or slow a person speaks. These devices would benefit from use of a 
speed factor as an input to a time-scale modification algorithm and 
further use the speed factor to equalize the rate of speech. 
With respect to the aspect of paging involving time-scaling of voice 
signals and to other applications such as dictation and voice mail, 
current methods of time-scaling lack the ideal combinations of providing 
adequate speech quality and flexibility that allows a designer to optimize 
the application within the constraints given. Thus, there exists a need 
for a voice communication system that is economically feasible and 
flexible in allowing optimization within a given configuration, and more 
particularly with respect to paging applications, that further retains 
many of the advantages of Motorola's high speed paging protocols. 
SUMMARY OF THE INVENTION 
In accordance with a first aspect of the present invention, a communication 
system using voice compression has at least one transmitter base station 
and a plurality of selective call receivers. The at least one transmitter 
base station comprises an input device which receives an audio voice 
message; a memory which stores the audio voice message; a processing 
device which digitizes the audio voice message to provide an input signal 
for analysis, divides the input signal into a sequence of equivalent time 
frames, for determining differences in a short-term frequency spectrum 
between adjacent time frames to derive a sequence of distance 
measurements, computes a first speed factor as an average of the sequence 
of distance measurements, and time-scales the input signal in accordance 
with the first speed factor to provide a time-scaled signal; and a 
transmitter which transmits the time-scaled signal The plurality of 
selective call receivers comprise a selective call receiver which receives 
the time-scaled signal, a processing device which demodulates and expands 
the time-scaled signal in accordance with the first speed factor to 
provide a reconstructed signal, and an amplifier which amplifies the 
reconstructed signal into a reconstructed audio signal. 
In accordance with a second aspect of the present invention, a transmitting 
base station comprises an input device which receives an audio voice 
message; a memory which stores the audio voice message; a processing 
device which digitizes the audio voice message to provide an input signal 
for analysis, divides the input signal into a sequence of equivalent time 
frames, determines differences in a short-term frequency spectrum between 
adjacent time frames to derive a sequence of distance measurements, 
computes a first speed factor as an average of the sequence of distance 
measurements, and time-scales the input signal in accordance with the 
first speed factor to provide a time-scaled signal; and a transmitter 
which transmits the time-scaled signal.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
The present invention provides a method to detect whether a speaker is 
speaking fast or slow and come up with an optimal compression factor to 
provide a certain quality of reconstructed speech when used with 
time-scaling techniques. There are several parameters or factors that 
could be used to determine how fast or how slow a speaker is speaking. The 
first one is the rate at which the short-term frequency spectrum changes 
in the speech. The second parameter or factor is the lengths of pauses 
between words. These two factors are used to determine a speed factor 
which determines how fast or slow a speaker talks. Optionally, a third 
factor would be the approximate pitch determined from a particular speaker 
to improve the accuracy of the speed factor. 
The system comprises of digitizing speech and then preferably dividing them 
up into frames of 20 ms. Linear Predictive Analysis (LPC analysis) is 
performed on each frame of speech and the difference in the short-term 
frequency spectrum between adjacent blocks is determined by using the 
Itakura distance measure D which is represented by the formula: 
##EQU1## 
"R" is the Auto-correlation matrix related to a frame n, "a" is the LPC 
vector related to frame n and "b" is the LPC vector related to frame n+1. 
D is the distance measure between frame n and n+1. The larger the distance 
D the greater the difference in frequencies between adjacent frames. An 
overall average measure of distance is then calculated for the whole 
speech sentence. This gives a measure of the speed factor. 
Next the energy and zero crossings are measured for each frame and silence 
periods are detected. The percentage of silence periods in the speech 
sentence gives another measure of the speed factor. These two parameters 
(and optionally, a pitch estimation parameter) are used to then determine 
the right compression factor that needs to be used along with time-scaling 
techniques so as to achieve a given quality of reconstructed speech. 
A brief description of a sample pitch estimation procedure is given below. 
1) Frame input speech into 20 ms blocks. 
2) Compute energy in each block. 
3) Compute average energy per block. 
4) Determine energy threshold to detect voiced speech as a function of the 
average energy per block. 
5) Using the energy threshold determine contiguous blocks of voiced speech 
of a length of at least 5 blocks. 
6) On each block of the contiguous voice speech found in step 5, do a pitch 
analysis. This could be done using a variety of methods including Modified 
Auto correlation method, AMDF or Clipped auto correlation method. 
7) The pitch values are smoothened using a median filter to eliminate 
errors in the estimation. 
8) Average all the smoothened pitch values to obtain an approximate 
estimate of the speaker's pitch. 
A sampling rate of 8 Khz is assumed in all cases above although other 
sampling rates and other methods of pitch estimation are contemplated 
within the present invention. 
As will be shown with respect to FIG. 3, S.sub.1 and S.sub.2 are the speed 
factors determined using the Itakura distance measure D and silence 
periods respectively. They are in turn (optionally, along with the pitch 
estimate) used to determine the final compression and expansion factor 
.alpha.. Thus, the Itakura distance measure and Silence period are used to 
determine the speed factor for a given speech sentence and further used to 
compute an optimum compression/expansion factor for a given quality of 
reconstructed speech. 
Referring to FIG. 1, a communication system illustrative of the voice 
compression and expansion techniques of the present invention are shown in 
a block diagram of the selective call system 100 which comprises an input 
device for receiving an audio signal such as telephone 114 from which 
voice based Selective calls are initiated for transmission to selective 
call receivers in the system 100. Each selective call entered through the 
telephone 114 (or other input device such as a computer) typically 
comprises (a) a receiver address of at least one of the selective call 
receivers in the system and (b) a voice message. The initiated selective 
calls are typically provided to a transmitter base station or a selective 
call terminal 113 for formatting and queuing. Voice compression circuitry 
101 of the terminal 113 serves to compress the time length of the provided 
voice message (the detailed operation of such voice compression circuitry 
101 is further discussed in the following description of FIGS. 2-5). 
Preferably, the voice compression circuitry 101 includes a processing 
device for compressing the audio signal using a time-scaling technique and 
a single sideband modulation technique to provide a processed signal, 
although other modulation techniques are contemplated for use in the 
present invention. The selective call is then input to the selective call 
transmitter 102 where it is applied as modulation to a radio frequency 
signal which is sent along with a time-scale factor over the air through 
an antenna 103. Preferably, the transmitter is a quadrature amplitude 
modulation transmitter for transmitting the processed signal, although 
other transmitters are contemplated for use in the present invention. 
An antenna 104 within a selective call receiver 112 receives the modulated, 
transmitted radio frequency signal and inputs it to a selective call 
receiver module or radio frequency receiver module 105 for receiving the 
processed signal or radio frequency signal, where the radio frequency 
signal is demodulated and the receiver address and the single side band 
(QAM) signal are recovered. This signal is then provided to an analog to 
digital converter (A/D) 115. Preferably, the selective call receiver 112 
includes a processing device for demodulating the received processed 
signal using a single sideband demodulation technique and a time-scaling 
expansion technique to provide a reconstructed signal. The compressed 
voice message is then provided to a voice expansion circuit 106 where the 
time length of the voice message is preferably expanded to the desired 
value. The voice message is then provided to a digital-to-analog converter 
(not shown) and an amplifier such as audio amplifier 108 for the purpose 
of amplifying it to a reconstructed audio signal. 
The demodulated receiver address and the time-scale factor are supplied 
from the radio frequency receiver 105 to a decoder 107. If the receiver 
address matches any of the receiver addresses stored in the decoder 107, 
an alert 111 is optionally activated, providing a brief sensory indication 
to the user of the selective call receiver 112 that a selective call has 
been received; The brief sensory indication may comprise an audible 
signal, a tactile signal such as a vibration, or a visual signal such as a 
light, or a combination thereof. The amplified voice message is then 
furnished from the audio amplifier 108 to an audio loudspeaker within the 
alert 111 for message announcement and review by the user. 
The decoder 107 may comprise a memory in which the received voice messages 
can be stored and recalled repeatedly for review by actuation of one or 
more controls 110. 
In another aspect of the invention, portions of FIG. 1 can be equally 
interpreted as part of a dictation device, voice mail system, answering 
machine, or sound track editing device for example. By removing the 
wireless aspects of the system 100 including the removal of selective call 
transmitter 102 and radio frequency receiver 105, the system can be 
optionally hardwired from the voice compression circuitry 101 to the voice 
expansion circuitry 106 through the A/D 115 as shown with the dashed line. 
Thus, in a voice mail, answering machine, sound track editing or dictation 
system, an input device 114 would supply an acoustic input signal such as 
a speech signal to the terminal 113 having the voice compression circuitry 
101. The voice expansion circuitry 106 and controls 110 would supply the 
means of listening and manipulating to the output speech signal in a voice 
mail, answering machine, dictation, sound track editing or other 
applicable system. This invention clearly contemplates that the 
time-scaling techniques of the claimed invention has many other 
applications besides paging. The paging example disclosed herein is merely 
illustrative of one of those applications. 
Referring to FIG. 2, the base station and transmitting portion 200 of the 
present invention is shown in FIG. 2. In operation, incoming voice 
messages are received via telephone line inputs 203, 205, 207, are 
processed by standard telephone-line interface circuits 202 (preferably 
including a PBX system) and then temporarily recorded, either on tape or 
in an electronic memory, i.e., in a buffer memory. The standard telephone 
interface and buffer memory portions in block 202 are referred to as a 
voice store and forward system and is described in detail in Motorola 
Instruction Manual No. 68P81105C25, entitled, "Voice Store and Forward 
Modax Plus/Metro-Page". 
Then, the messages are retrieved from the buffer memory within block 202 
and are processed by the speech detector block 206 which preferably 
includes LPC analysis circuits that analyze the voice signal and generate 
the digital LPC parameters that correspond to the input voice signal. 
The speech rate detector 206 computes and provides a time scaling factor 
.alpha. to a Compression Time Scaling Block 208 which compresses the input 
speech signal in accordance with the time scaling factor. The Block 208 is 
coupled to the controller 210 and may optionally include memory block 209. 
In speech system of the present invention, the intent is to provide a 
technique in which several speech messages can be transmitted in the time 
presently required to transmit one message while minimizing the affects on 
speech quality. 
As mentioned above, the speech signals from the phone line inputs 203, 205, 
and 207 are stored temporarily in a storage memory within block 202. This 
is done to handle high peak input loads without requiring a large number 
of expensive analyzers. The stored voice samples are retrieved from the 
memory under the direction of the controller 210 which, in practice can be 
a microcomputer. 
FIG. 3 illustrates a further detailed block diagram of the speech rate 
detector 206 of FIG. 2. The speech rate detector 206 preferably comprises 
several analysis steps that ultimately results in more efficient 
compression or expansion with an optimal level of desired speech quality. 
The detector 206 preferably comprises an LPC analyzer 250 that provides an 
LPC analysis on the input frames of the input signal. Preferably, as 
previously discussed, the input signal is digitized and divided into 
frames of 20 milliseconds and stored in a memory (see memory 209, FIG. 2). 
At block 252, the Itakura distance measure D is determined between frames. 
In Block 254, the average distance using the distance D from each frame 
for a whole speech signal (for example, from an entire sentence comprising 
a message), is computed preferably from one source or speaker. As a result 
of the average distance computed and determining a difference in the 
short-term frequency spectrum between adjacent time frames in block 254, 
an overall distance measure and as a result a first speed factor S.sub.1 
is provided to the final decision block 258. The final decision block 258 
can thus provide a time-scale factor .alpha. to appropriately compress (or 
expand) the input speech signal in accordance with the time-scaling 
factor. Depending on the pitch estimate of the speaker, the Itakura 
distances can be used to compress voice input signals in accordance with 
the following chart: 
______________________________________ 
ITAKURA DISTANCE RANGE 
*Pitch &lt; 40 samples 
*Pitch &gt; 40 Samples 
COMPRESSION RATE 
______________________________________ 
-0.4 to -0.45 
&lt;-0.3 4 
-0.45 to -0.5 
-0.35 to -0.4 3.5 
-0.5 to -0.55 
-0.4 to -0.45 
3.25 
&lt;-0.55 -0.45 to -0.5 3 
&lt;-0.5 2.5 
______________________________________ 
*Samples at 8 kilohertz sampling rate. 
The values above can vary for optimization in different systems. Such 
factors as language, number of users, the ratio between genders among 
users of a system could be some of the many factors that may be used to 
alter the figures in the chart above to provide an optimum system. 
Further refinement of the method can be achieved using a second speed 
factor S.sub.2 by performing a silence analysis at block 260. In other 
words, the energy and zero crossings are measured for each frame and using 
this, silence periods are detected. In block 262, the percentage of 
silence periods in the speech input signal thus provides another measure 
and a second speed factor S.sub.2 which is computed in block 264. 
Therefore, in view of the graph above, if the silence is less than 10%, 
you can reduce the compression rate by 0.25. The value actually obtained 
from block 264 may be adjusted based on the level of quality of speech 
desired. Therefore, in this instance, the two speed factors S.sub.1 and 
S.sub.2 are provided to decision block 258 and used to then determine the 
right overall compression (or expansion) factor that needs to be used 
along with time-scaling techniques to achieve a given quality of 
reconstructed speech. Additionally, the present invention can optionally 
determine an average pitch estimate in block 256 to provide another factor 
to the decision block 258, allowing for further refinement in the decision 
to compress without audibly affecting the desired optimum quality of 
resultant speech. Average pitch estimates in this instance will aid in 
determining whether a male or female speaker is providing the input 
signal. This knowledge in combination with the previously determined speed 
factor will provide a time-scale factor which is used to obtain optimum 
quality reconstructed speech. 
FIG. 4 illustrates a block diagram of a first embodiment of a transmitter 
300 in accordance with the present invention. An analog speech signal is 
input to an anti-aliasing low pass filter 301 which strongly attenuates 
all frequencies above one-half the sampling rate of an analog-to-digital 
converter (ADC) 303 which is further coupled to the filter 301. The ADC 
303 preferably converts the analog speech signal to a digital signal so 
theft further signal processing can be done using digital processing 
techniques. Digital processing is the preferred method, but the same 
functions could also be performed with analog techniques or a combination 
of analog and digital techniques. 
A band pass filter 305 coupled to the ADC 303 strongly attenuates 
frequencies below and above its cutoff frequencies. The lower cutoff 
frequency is preferably 300 Hz which allows the significant speech 
frequencies to pass, but attenuates lower frequencies which would 
interfere with a pilot carrier. The upper cutoff frequency is preferably 
2800 Hz which allows the significant speech frequencies to pass but 
attenuates higher frequencies which would interfere with adjacent 
transmission channels. An automatic gain control (AGC) block 307 
preferably coupled to the filter 305 equalizes the volume level of 
different voices. The block 307 then provides a signal to the speech rate 
detector block 206, which operates as previously described with regard to 
FIG. 3 to provide a time-scaling factor to the time compression block 309. 
The time compression block 309 preferably coupled to the AGC block 307 
shortens the time required for transmission of the speech signal while 
maintaining essentially the same signal spectrum as at the output of the 
bandpass filter 305. The time compression method is preferably WSOLA-SD 
(as explained in U.S. patent application Ser. No. 08/395,739), but other 
methods could be used. An amplitude compression block 311, and the 
corresponding amplitude expansion block in a receiver (not shown), form a 
companding device which is well known to increase the apparent 
signal-to-noise ratio of the received speech. The companding ratio is 
preferably 2 to 1 in decibels, but other ratios could be used in 
accordance with the present invention. In the particular instance of a 
communication system such as a paging system, the devices 301-309 may be 
included in a paging terminal (113 of FIG. 1) and the remaining components 
in FIG. 4 could constitute a paging transmitter (102 of FIG. 1). In such a 
case, there would typically be a digital link between the paging terminal 
and paging transmitter. For instance, the signal after block 309 could be 
encoded using a pulse code modulation (PCM) technique and then 
subsequently decoded using PCM to reduce the number of bits transferred 
between the paging terminal and paging transmitter. 
In any event; a second band pass filter 308 coupled to the amplitude 
compression block 311 strongly attenuates frequencies below and above its 
cutoff frequencies to remove any spurious frequency components generated 
by the AGC 307, the time compression block 309 or the amplitude 
compression block 311. The lower cutoff frequency is preferably 300 Hz 
which allows the significant speech frequencies to pass, but attenuates 
lower frequencies which would interfere with the pilot carrier. The upper 
cutoff frequency is preferably 2800 Hz which allows the significant speech 
frequencies to pass but attenuates higher frequencies which would 
interfere with adjacent transmission channels. 
The time compressed speech samples are preferably stored in a buffer 313 
until an entire speech message has been processed. This allows the time 
compressed speech message to then be transmitted as a whole. This 
buffering method is preferably used for paging service (which is typically 
a non real time service). Other buffering methods may be preferable for 
other applications. For example, for an application involving two-way real 
time conversation, the delay caused by this type of buffering could be 
intolerable. In that case it would be preferable to interleave small 
segments of several conversations. For example, if the time compression 
ratio is 3:1, then 3 real time speech signals could be transmitted via a 
single channel. The 3 transmissions could be interleaved on the channel in 
150 millisecond bursts and the resulting delays would not be 
objectionable. The time compressed speech signal from the buffer 313 is 
applied both to a Hilbert transform filter 323 and to a time delay block 
315 which has the same delay as the Hilbert transform filter, but does not 
otherwise affect the signal. 
The output of the time delay block 315 (through the summing circuit 317) 
and the Hilbert transform filter 323 form, respectively, the in-phase (I) 
and quadrature (Q) components of an upper sideband (USB) single sideband 
(SSB) signal. The output of the time delay and the negative (325) of the 
Hilbert transform filter form, respectively, the in-phase (I) and 
quadrature (Q) components of a lower sideband (LSB) single sideband 
signal. Thus the transmission may be on either the upper or lower 
sideband, as indicated by the dotted connection. 
While the upper sideband is used to transmit one time compressed speech 
signal, the lower sideband can be used to simultaneously transmit a second 
time compressed speech signal by using another similar transmitter 
operating on the lower sideband. SSB is the preferred modulation method 
because of efficient use of transmission bandwidth and resistance to 
crosstalk. Double sideband Amplitude Modulation (AM) or frequency 
modulation (FM) could be used, but would require at least twice the 
bandwidth for transmission. It is also possible to transmit one time 
compressed speech signal directly via the I component and a second time 
compressed speech signal directly via the Q component, however, in the 
present embodiment this method is subject to crosstalk between the two 
signals when multipath reception occurs at the receiver. 
A direct current (DO) signal is added to the I component of the signal to 
generate the pilot carrier, which is transmitted along with the signal and 
used by the receiver to substantially cancel the effects of gain and phase 
variations or fading in the transmission channel. The I and Q components 
of the signal are converted to analog form by digital-to-analog converters 
(DAC) 319 and 327 respectively. The two signals are then filtered by low 
pass reconstruction filters 321 and 329 respectively to remove spurious 
frequency components resulting from the digital-to-analog conversion 
process. A quadrature amplitude modulation (QAM) modulator 333 modulates 
the I and Q signals onto a radio frequency (RF) carrier at low power 
level. Other modulation methods, e.g. direct digital synthesis of the 
modulated signal would accomplish the same purpose as the DACs (319 and 
327), reconstruction filters (321 and 329), and QAM modulator 333. 
Finally, a linear RF power amplifier 335 amplifies the modulated RF signal 
to the desired power level, typically 50 watts or more. Then, the output 
of the RF power amplifier 335 is coupled to the transmitting antenna. 
Other variations can produce essentially the same results. For example, 
the amplitude compression could be performed before the time compression, 
or omitted altogether and the device would still perform essentially the 
same function. 
FIG. 5 illustrates a block diagram of a second embodiment of a transmitter 
400 in accordance with the present invention. In FIG. 5, both the upper 
and lower sidebands are used to simultaneously transmit different portions 
of the same time compressed signal. The transmitter 400 preferably 
includes an anti-alias filter 404, an ADC 403, a bandpass filter 405, an 
AGC 407, a time compression block 409, an amplitude compression block 411, 
and a bandpass filter 408 coupled and configured as in FIG. 4. The block 
407 provides a signal to the speech rate detector block 206, which 
operates as previously described with regard to FIG. 3 to provide a 
time-scaling factor to the time compression block 409. Operation of the 
transmitter of FIG. 4 is the same as in FIG. 3 until an entire speech 
message has been processed and stored in a buffer 413. The time compressed 
speech samples stored in the buffer 413 are then divided to be transmitted 
on either the upper or lower sideband. Preferably, the first half of the 
time compressed speech message is transmitted via one sideband and the 
second half of the time compressed speech message is transmitted via the 
other sideband (or alternatively on each of the I and Q components 
directly). 
The first portion of time compressed speech signal from the buffer 413 is 
applied to both a first Hilbert transform filter 423 and to a first time 
delay block 415 which has the same delay as the Hilbert transform filter 
423 but does not otherwise affect the signal. The output of the first time 
delay (through summing circuit 417) and the first Hilbert transform filter 
423 (through summing circuit 465) are In-Phase (I) and Quadrature Phase 
(Q) signal components which, when coupled to I and Q inputs of the QAM 
modulator, generate upper sideband signal having information only from the 
first portion of time compressed speech samples. The second time 
compressed speech signal from the buffer 413 is applied to both a second 
Hilbert transform filter 461 and to a second time delay block 457 which 
has the same delay as the Hilbert transform filter 461 but does not 
otherwise affect the signal. The output of the second time delay (through 
summing circuits 459 and 417) and the negative (463) of the output of the 
second Hilbert transform filter 461 (and again, through summing circuit 
465) are In-Phase (I) and Quadrature Phase (Q) signal components which, 
when coupled to I and Q inputs of the QAM modulator, generate upper 
sideband signal having information only from the second portion of time 
compressed speech samples. The I components of the upper and lower 
sideband signals are added with a DC pilot carrier component (through 
summing circuit 459) to form a composite I component for transmission. The 
Q components of the upper and lower sideband signals are added (through 
summing circuit 465) to form a composite Q component for transmission. It 
will be appreciated that elements 415, 423, 457, 461, 417, 459, 463, 465, 
419, 427, 421, and 429 form a preprocessor which generates preprocessed I 
and Q signal components, which when coupled to the QAM modulator 453 
generate the low level subchannel signal with a subcarrier F.sub.A, having 
two single sideband signals which have independent information on each 
sideband. 
The transmitter 400 further comprises DACs 419 and 427, reconstruction 
filters 421 and 429, QAM modulator 433, and RF power amplifier 455 
arranged and constructed as described in FIG. 4. Operation of the rest of 
the transmitter of FIG. 5 is the same as in FIG. 4. 
Preferably, in both transmitters 300 and 400 of FIGS. 4 and 5 respectively, 
only the anti-alias filters, the reconstruction filters, the RF power 
amplifier and optionally the Analog to Digital converter and digital to 
analog converters are separate hardware components. The remainder of the 
devices can preferably be incorporated into software which could be run on 
a processor, preferably a digital signal processor.