Note : Les descriptions sont présentées dans la langue officielle dans laquelle elles ont été soumises.
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REPETITIVE.SOUND COMPRESSION SYSTEM
' Field of the Invention
The present invention teaches a system for
' S compressing quasi-periodic sound by comparing it to
presampled portions in a codebook.
Background and Summar
Many sound compression schemes take advantage of
the repetitive nature of everyday sounds. For example,
the standard human voice coding device or "vocoder", is
often used for compressing and encoding human voice
sounds. A vocoder is a class of voice coder/decoders
that models the human vocal tract.
A typical vocoder models the input sound as two
parts: the voice sound known as V, and the unvoice sound
known as U. The channel through which these signals are
conducted is modelled as a lossless cylinder. The output
speech is compressed based on this model.
Strictly speaking, speech is not periodic.
However, the voice part of speech is often labeled as
quasi-periodic due to its pitch frequency. The sounds
produced during the un-voiced region, are highly random.
Speech is always referred to as non-stationary and
stochastic. Certain parts of speech may have redundancy
and perhaps correlated to some prior portion of speech to
some extent, but they are not simply repeated.
The main intent of using a vocoder is to find ways
to compress the source, as opposed to performing
compression of the result. The source in this case is
the excitation formed by glottal pulses. The result is
the human speech we hear. However, there are many ways
that the human vocal tract can modulate the glottal
pulses to form human voice. Estimations of the glottal
pulses are predicted and then coded. Such a model
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reduces the dynamic range of the resulting speech, hence
rendering the speech more compressible.
More generally, the special kind of speech '
filtering can remove speech portions that are not
perceived by the human ear. With the vocoder model in '
place, a residue portion of the speech can be made
compressible due to its lower dynamic range.
The term "residue" has multiple meanings. It
generally refers to the output of the analysis filter,
ZO the inverse of the synthesis filter which models the
vocal tract. In the present situation, residue takes on
multiple meanings at different stages: at stage 1- after
the inverse filter (all zero filter), stage 2: after the
long term pitch predictor or the so-called adaptive pitch
VQ, stage 3: after the pitch codebook, and at stage 4:
after the noise codebook. The term "residue" as used
herein literally refers to the remaining portion of the
speech by-product which results from previous processing
stages.
The preprocessed speech is then encoded. A
typical vocoder uses an 8 kHz sampling rate at Z6 bits
per sample. These numbers are nothing magic, however -
they are based on the bandwidth of telephone lines.
The sampled information is further processed by a
speech codec which outputs an 8 kHz signal. That signal
may be post-processed, which may be the opposite of the
input processing. Other further processing that is
designed to further enhance the quality and character of
the signal may be used.
The suppression of noise also models the way that
humans perceives sound. Different weights are used at
different times according to the strength of speech both
in the frequency and time domain. The masking properties
of human hearing cause loud signals at different
frequencies to mask the effect of low level signals
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around those frequencies. This is also true in the time
domain. The result is that more noise can be tolerated
during that portion of time and frequency. This allows
us to pay more attention elsewhere. This is called
' 5 "perceptual weighting" - it allows us to pick vectors
which are perpectually more effective.
The human vocal tract can be (and is) modeled by a
set of lossless cylinders with varying diameters.
Typically, it a.s modeled by an 8 to 12th order all-pole
filter 1/A(Z). Its inverse counterpart A(Z) is an all-
zero filter with the same order. Output speech is
reproduced by exciting the synthesis filter 1/A(z) with
the excitation. The excitation, or glottal pulses is
estimated by inverse filtering the speech signal with the
inverse filter A(z). A digital signal processor often
models the synthesis filter as the transfer function H(V)
- 1/A(z). This means that this model is an all-pole
process. Ideally, the model is more complicated, and
includes both poles and zeros.
Much of the compressibility of speech comes from
its quasi-periodicity. Speech is quasi-periodic due to
its pitch frequency around voice sound. Male speech
usually has a pitch between 50 and 100 Hz. Female speech
usually has a pitch above 100 Hz.
While the above describes compression systems for
voice coding, the same general principles are used to
code and compress other similar kinds of sound.
Various techniques are known for improving the
model. Each of these techniques, however, increases the
necessary bandwidth to carry the signal. This produces a
tradeoff between bandwidth of the compressed signal and
quality of the non=steady-state sound.
These problems are overcome according to the
present invention by new features.
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A first aspect of the present invention includes a
new architecture for coding which allows various coding
and monitoring advantages. The disclosed system of the '
present invention includes new kinds of codebooks for
coding. These new codebooks allow faster consequence to '
changes in the input sound stream. Essentially, these
new codebooks use the same software routine many times
over, to improve coding efficiency.
Brief Description of the Drawings
These and other aspects of the present invention
will now be described with reference to the accompanying
drawings in which:,
Figure 1 shows a block diagram of the basic
vocoder of the present invention; and
Figure 2 the advanced codebook technique of the
present invention.
~escrit~tion of the Preferred Embodiments
Figure 1 shows the advanced vocoder of the present
invention. The current speech codec uses a special class
of vocoder which operates based on LPC (linear
predictive coding). All future samples are being
predicted by a linear combination of previous samples and
the difference between predicted samples and actual
samples. As described above, this is modeled after a
lossless tube also known as an all-pole model. The model
presents a relative reasonable short term prediction of
speech.
The above diagram depicts such a model, where the
input to the lossless tube is defined as an excitation
which is further modeled as a combination of periodic
pulses and random noise.
A drawback of the above model is that the vocal
tract does not behave exactly as a cylinder and is not
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lossless. The human vocal tract also has side passages
such as the nose.
' Speech to be coded 100 is input to an analysis
block 102 which analyzes the content of the speech as
described herein. The analysis block produces a short
term residual alone with other parameters.
Analysis in this case refers as LPC analysis as
depicted above in our lossless tube model, that includes,
for example, computation of windowing, autocorrelation,
Durbin's recursion, and computation of predictive
coefficients are performed. In addition, filtering
incoming speech with the analysis filter based on the
computed predictive coefficients will generate the
residue, the short term residue STA res 104.
This short term residual 104 is further coded by
the coding process 110, to output codes or symbols 120
indicative of the compressed speech. Coding of this
preferred embodiment involves performing three codebook
searches, to minimize the perceptually-weighted error
signal. This process is done in a cascaded manner such
that codebook searches are done one after another.
The current codebooks used are all shape gain VQ
codebooks. The perceptually-weighted filter is generated
adaptively using the predictive coefficients from the
current sub-frame. The filter input is the difference
between the residue from previous stage versus the shape
gain vector from the current stage, also called the
residue, is used for next stage. The output of this
filter is the perceptually weighted error signal. This
operation is shown_and explained in more detail with
reference to Figure 2. Perceptually-weighted error from
each stage is used as a target for the searching in next
stage.
' The compressed speech or a sample thereof 122 is
also fed back to a synthesizer 124, which reconstitutes a
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reconstituted original block 126. The synthesis stage
decodes the linear combination of the vectors to form a
reconstruction residue, the result is used to initialize "
the state of the next search in next sub-frame.
Comparison of the original versus the
reconstructed sound results in an error signal which will
drive subsequent codebook searches to further minimize
such perceptually-weighted error. The objective of the
subsequent coder is to code this residue very
efficiently.
The reconstituted block 126 indicates what would
be received at the~receiving end. The difference between
the input speech 100 and the reconstituted speech 126
hence represents an error signal 132.
This error signal is perceptually weighted by
weighting block 134. The perceptual weighting according
to the present invention weights the signal using a model
of what would be heard by the human ear. The
perceptually-weighted signal 136 is then heuristically
processed by heuristic processor 140 as described herein.
Heuristic searching techniques are used which take
advantage of the fact that some codebooks searches are
unnecessary and as a result can be eliminated. The
eliminated codebooks are typically codebooks down the
search chain. The unique process of dynamically and
adaptively performing such elimination is described
herein.
The selection criterion chosen is primarily based
on the correlation_between the residue from a prior stage
versus that of the current one. If they are correlated
very well, that means the shape-gain vQ contributes very
little to the process and hence can be eliminated. On
the other hand, if it does not correlate very well the
contribution from the codebook is important hence the "
index shall be kept and used.
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Other techniques such as stopping the search when
an adaptively predetermined error threshold has been
' reached, and asymptotic searches are means of speeding up
the search process and settling with a sub-optimal
result. The heuristically-processed signal 138 is used
as a control for the coding process 110 to further
improve the coding technique.
This general kind of filtering processing is well
known in the art, and it should be understood that the
present invention includes improvements on the well known
filtering systems in the art.
The coding according to the present invention uses
the codebook types and architecture shown in Figure 2.
This coding includes three separate codebooks: adaptive
vector quantatization (VQ) codebook 200, real pitch
codebook 202, and noise codebook 204. The new
information, or residual 104, is used as a residual to
subtract from the code vector of the subsequent block.
ZSR (Zero state response) is a response with zero input.
The ZSR is a response produced when the code vector is
all zeros. Since the speech filter and other associated
filters are IIR (infinite impulse response) filters, even
when there is no input, the system will still generate
output continuously. Thus, a reasonable first step for
codebook searching is to determine whether it is
necessary to perform any more searches, or perhaps no
code vector is needed for this subframe.
To clarify this point, any prior event will have a
residual effect. Although that effect will diminish as
time passes, the effect is still present well into the
next adjacent sub-frames or even frames. Therefore, the
' speech model must take these into consideration. If the
speech signal present in the current frame is just a
residual effect from a previous frame, then the
perceptually-weighted error signal Eo will be very low or
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even be zero. Note that, because of noise or other
system issues, all-zero error conditions will almost
never occur.
eo = STA res - ~. The reason ~ vector is used is '
for completeness to indicate zero state response. This
is a set-up condition for searches to be taken place. If
E~ is zero, or approaches zero, then no new vectors are
necessary.
EO is used-to drive the next stage as the ~~target~~
of matching for the next stage. The objective is to find
a vector such that E1 is very close to or equal to zero,
where E1 is the perceptually weighted error from el, and
e1 is the difference between e0-vector(i). This process
goes on and on through the various stages.
The preferr-ed mode of the present invention uses a
preferred system with 240 samples per frame. There are
four subframes per frame, meaning that each subframe has
60 samples.
A VQ search for each subframe is done. This VQ
search involves matching the 60-part vector with vectors
in a codebook using a conventional vector matching
system.
Each of these vectors must be defined according to
an equation. The basic equation used is of the form that
2 5 G$Ai + GbBj + G~Ck.
Since the objective is to come up with a minimum
perceptually weighted error signal E3 by selecting
vectors Ai, Bj, and Ck along with the corresponding gain
Ga, Gb, and Gc. This does NOT imply the vector sum of
3 0 Ga"Ai + GbBj + G~CE = STA_res .
In fact, it is almost never true with the
exception of silence.
The error value Eo is preferably matched to the
values in the AVQ codebook 200. This is a conventional
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kind of codebook where samples of previous reconstructed
speech, e.g., the last 20 ms, is stored. A closest match
' is found. The value el (error signal number 1? represents
the leftover between the matching of Eo with AVQ 200.
According to the present invention, the adaptive
vector quantizer stores a 20 ms history of the
reconstructed speech. This history is mostly for pitch
prediction during voice frame. The pitch of a sound
signal does not change quickly. ,The new signal will be
20 closer to those values in the AVQ than they will to other
things. Therefore, a close match is usually expected.
Changes in voice, however, or new users entering a
conversation, will degrade the quality of the matching.
According to the present invention, this degraded
matching is compensated using other codebooks.
The second codebook used according to the present
invention is a real pitch codebook 202. This real pitch
codebook includes code entries for the most usual
pitches. The new pitches represent most possible pitches
of human voices, preferably from 200 Hz down. The
purpose of this second codebook is to match to a new
speaker and for startup/voice attack purposes. The pitch
codebook is intended for fast attack when voice starts or
when a new person entering the room with new pitch
information not found in the adaptive codebook or the so-
called history codebook. Such a fast attack method
allows the shape of speech to converge more quickly and
allows matches more closely to that of the original
waveform during the voice region.
Usually when a new speaker enters the sound field,
AVQ will have a hard time performing the matching.
Hence, El is still very high. During this initial time,
therefore, there are large residuals, because the
matching in the codebook is very poor. The residual El
represents the new speaker's pitch weighted error. This
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residual is matched to the pitch in the real pitch
codebook 202.
The conventional method uses some form of random '
pulse codebook which is slowly shaped via the adaptive
process in 200 to match that of the original speech. °
This method takes too long to converge. Typically it
takes about 6 sub-frames and causes major distortion
around the voice attack region and hence suffers quality
loss.
The inventors have found that this matching to-the
pitch codebook 202 causes an almost immediate re-locking
of the signal. For example, the signal might be re-
locked in a single period, where one sub-frame period =
60 samples = 60/8000 = 7.5ms. This allows accurate
representation of the new voice during the transitional
period in the early part of the time while the new
speaker is talking.
The noise codebook 204 is used to pick up the
slack and also help shape speech during the unvoiced
period.
As described above, the G's represent amplitude
adjustment characteristics, and A, B and C are vectors.
The codebook for the AVQ preferably includes 256
entries. The codebooks for the pitch and noise each
include 512 entries.
The system of the present invention uses three
codebooks. However, it should be understood that either
the real pitch codebook or the noise codebook could be
used without the other.
Additional processing is carried out according to
the present invention under the characteristic called
heuristics. As described above, the three-part codebook
of the present invention improves the efficiency of
matching. However, this of course is only done at the
expense of more transmitted information and hence less
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compression efficiency. Moreover, the advantageous
architecture of the present invention allows viewing and
processing each of the error values eo-e3 and Eo-E3. These
error values tell us various things about the signals,
including the degree of matching. For example, the error
value Eo being 0 tells us that no additional processing is
necessary. Similar information can be obtained from
errors Eo-E3. According to the present invention, the
system determines the degree of mismatching to the
codebook, to obtain an indication of whether the real
pitch and noise codebooks are necessary. Real pitch and
noise codebooks are not always used. These codebooks are
only used when some new kind or character of sound enters
the field.
I5 The codebooks are adaptively switched in and out
based on a calculation carried out with the output of the
codebook.
The preferred technique compares Eo to E1. Since
the values are vectors, the comparison requires
correlating the two vectors. Correlating two vectors
ascertains the degree of closeness therebetween. The
result of the correlation is a scalar value that
indicates how good the match is. If the correlation
value is low, it indicates that these vectors are very
different. This implies the contribution from this
codebook is significant, therefore, no additional
codebook searching steps are necessary. On the contrary,
if the correlation value is high, the contribution from
this codebook is not needed, then further processings are
required. Accordingly, this aspect of the invention
compares the two error values to determine if additional
codebook compensation is necessary. If not, the
additional codebook compensation is turned off to
increase the compression.
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A similar operation can be carried out between El
and EZ to determine if the noise codebook is necessary.
Moreover, those having ordinary skill in the art
will understand that this can be modified other ways
using the general technique that a determination of
whether the coding is sufficient is obtained, and the
codebooks are adaptively switched in or out to further
improve the compression rate and/or matching.
Additional heuristics are also used according to
the present invention to speed up the search. Additional
heuristics to speed up codebook searches are:
a) a subset of codebooks is searched and a partial
perceptually weighted error Ex is determined. If Ex is
within a certain predetermined threshold, matching is
stopped and decided to be good enough. Otherwise we
search through the end. Partial selection can be done
randomly, or through decimated sets.
b) An asymptotic way of computing the perceptually
weighted error is used whereby computation is simplified.
c) Totally skip the perceptually weighted error
criteria and minimize "e" instead. In such case, an
early-out algorithm is available to further speed up the
computation.
Another heuristic is the voice or unvoice
detection and its appropriate processing. The
voice/unvoice can be determined during preprocessing.
Detection is done, for example, based on zero crossings
and energy determinations. The processing of these
sounds is done differently depending on whether the input
sound is voice or unvoice. For example, codebooks can be
switched in depending on which codebook is effective.
Different codebooks can be used for different
purposes, including but not limited to the well known
technique of shape gain vector quantatization and join
optimization. An increase in the overall compression
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rate is obtainable based on preprocessing and switching
in and out the codebooks.
Although only a few embodiments have been
described in detail above, those having ordinary skill in
the art will certainly understand that many modifications
are possible in the preferred embodiment without
departing from the teachings thereof.
All such modifications are intended to be
encompassed within the following claims.
