A Simple LPC Vocoder
a given model order. Fortunately, most
Bob Beauchaine
EE586, Spring 2004
Linear Prediction Coefficient voice
coding/decoding attempts to take advantage
of the fact that sampled speech has
significant correlation between samples
(assuming sufficient sampling frequency).
Any signal which possesses this property is
amenable to linear prediction, in which
current speech samples are modeled as a
linear combination of past samples.
Minimization of the Mean Squared Error
between the predicted samples and their
actual values leads to a unique set of
coefficients for a chosen filter order. Thus
an LPC vocoder is at its heart little more
than a classical MMSE system.
Vocal tract modeling
Of course, any system can be
modeled in this sense as long as certain
Figure 1 – Human vocal tract acoustic model
conditions are met by the input signal. This
does not imply that the model is a
particularly “good” fit to the real data – just
that it is the best fit that can be achieved for
modern speech production models of the
human vocal tract consider the speech
production process to be sufficiently well
modeled as the output of a linear, time-
varying system excited by quasi-periodic
impulses, broadband random noise, or some
combination of the two.
Most tractable models of human
speech production consider the human vocal
tract to be a series of time-varying tubes of
various lengths and cross sections (see
Figure 1). The excitation source produces
input to the tube at one end, and the sound
propagates through the series of tubes,
bouncing off the walls, reflecting at the
junctions of each new section, and being
shaped by the instantaneous nature of
transmission path. Note that this simplified
view of speech production produces an all-
pole model. How well this assumption holds
up in reality has profound implications on
the quality of speech produced with its use,
which we will discuss later.
If the vocal tract can be modeled in
this fashion, we then need a model of the
excitation source to complete our view of
linear speech production. Analysis of
speech waveforms have shown that, in the
broadest of terms, speech can be broken
down into two major categories. The first
type of speech is called voiced speech. It is
produced by the periodic excitation of the
vocal tract at the larynx via vibration of the
vocal cords excited by the passage of air
expelled from the lungs (this excitation
source is generally referred to as a glottal
pulse). Voiced speech is characterized by a
pseudo-stationary primary frequency (called
a formant) plus harmonics. Voiced speech
is typically associated with sounds produced
by an “open” vocal tract – in English, vowels
provide the best example of voiced speech.
The other primary form of speech is
called unvoiced. Unvoiced speech is
produced by turbulence when air is forced
through a constriction in the vocal tract – by
pursed lips, tongue against teeth, or other
combinations. Unvoiced speech closely
resembles random noise, showing little
periodicity and little correlation between
samples. Of course, nothing as complex as
speech production can be reduced to a
binary model without some compromise.
Much more robust models segment speech
into many more categories, such as
Sound ‘aah’, male
speaker
6 5
5
4
1
0
-1
0 50 100 150 200 250 300
Red = spectrum of original speech
Blue = impulse response of 12 pole predicted
filter
Figure 2 LPC modeling of voiced/unvoiced speech
sonorants, voiced consonants, nasals, semi-
vowels, fricatives – all of which possess
quantifiable differences that could in theory
be used to improve LPC based speech
analysis and production, but which in reality
only manage to complicate the analysis.
Figure 2 LPC modeling of voiced/unvoiced
speech shows a comparison of the spectrum
of a segment of voiced and unvoiced speech
(above) and the impulse response of a 12th
order LPC filter predicted from that speech
(below). A cursory analysis shows that the
fine details of the speech are clearly lost,
but that the general envelope of the spectra
is retained. It is this ability of the LPC
model to track the gross spectral
characteristics of the human vocal tract that
makes is such a successful speech coding
model.
LPC Vocoder
Now that we have a rudimentary
understanding of speech production, we
may finally discuss the basic LPC speech
encoder/decoder system (see Figure 3 – LPC
encoder block diagram). First, speech is
sampled at a frequency appropriate to
capture all of the necessary frequency
components important for processing and
recognition. For voice transmission, 10kHz
is typically the sampling frequency of choice,
Sound ‘shhh’, male
speaker
4
3
1
0
-1
0 50 100 150 200 250 300
though 8kHz is not unusual. This is
because, for almost all speakers, all
significant speech energy is contained in
those frequencies below 4kHz (although
some women and children violate this
assumption). The speech is then
segmented into blocks for processing.
Simple LPC analysis uses equal length blocks
of between 10 and 30ms. Less than 10ms
does not encompass a full period of some
low frequency voiced sounds for male
speakers. My own experiments with male
speech sounded synthetic at 10ms sample
windows when, for certain frames, pitch
detection became impossible. More than
30ms violates the basic principle of
Figure 3 – LPC encoder block diagram
stationarity upon which the least squares
method relies.
Once segmented, the speech type is
determined as either voiced, unvoiced, or
silence. This simple sounding task is indeed
the most difficult issue in LPC vocoders –
the past 25 years have seen dozens of
papers published on varying methods of
accomplishing this feat reliably across
Figure 4 LPC decoder
speakers and environments.
If the speech is classified as voiced,
then its pitch must be determined. This is
because the human ear, while tolerant of
many errors in speech coding, is somewhat
sensitive to errors in pitch. Speech that is
produced with pitch errors is quite annoying
and typically sounds synthetic. Once the
type of speech is determined, the LPC
parameters are estimated. Crucial to the
speech production model is the prediction
order – the number of taps used in the
estimation process. Considerable analysis of
the prediction error has been examined in
the literature. It has been shown that the
normalized error drops steeply for prediction
orders from 0 to 4, then gradually reduces
up to order 12, thereafter flattening out into
a region of diminishing returns1. Thus, since
LPC systems are almost universally used in
low bit-rate applications, and since using a
prediction order of greater than 14 produces
little improvement, LPC coders most often
use a model of order 10-14.
In reality, the determination of LPC
parameters can be performed at any point
in the analysis process, and in fact some
V/U/S detectors make use of the LPC
parameters.
Once the LPC
parameters are
determined, the
energy of the
signal segment
is determined.
This is required
at the synthesis
end to equalize
energy levels of
the synthesized
speech. Again,
the energy
content of the
signal is, as we shall see, a very powerful
discriminator of speech type.
At the end of this process, we have
a model of the speech segment that
determines its type, energy content, pitch,
and LPC parameters. If it is not clear at this
point, it should be stated that, for an LPC
1
Digital Processing of Speech Signals, L.R.
Rabiner, R.W. Schafer, pg. 427.
vocoder, no portion of the actual speech
waveform is transmitted. The model is
sufficient for the receiver to synthesize N  M  M
1 1
speech at the receiver end that can be a
remarkable facsimile of the original. This ∑ a i⋅ 
M ∑
s ( k − i) s ( k − j)
M
⋅ ∑ s ( k) s ( k − j)

allows LPC vocoders to achieve significant i=1 k = 1  k =1

bit rate reduction – commercial systems are
available that code down to 2.4kbps2. where j varies from 1..N. By letting
At the receiver (see figure 4) the
transmitted model parameters are decoded  M 1 
to produce a time varying vocal tract inverse
filter. Depending on the V/U/S classification
φ ij 
M ∑
s ( k − i) s ( k − j)
made by the encoder, the excitation source k = 1 
is chose as being either a random noise
source or a periodic impulse train whose the “covariance” term be represented as
period was also transmitted by the encoder. we can write, in matrix form, the solution to
The excitation source is scaled appropriately the simultaneous set of LPC equations as
by the transmitted gain, and passed through
the inverse filter to produce synthetic
speech.
ΦA ψ
Because of the similarity to a covariance
Computations function, this method of LPC analysis is
One should never lose sight of the known as the covariance method.
fact that LPC is an attempt to significantly An alternative formulation of the LP
reduce, in real time, the bit rate required to equations proceeds as follows. First, the
transmit speech. Hence some discussion of limits on the sums for the analysis window
the LPC parameters is merited. If the are changed to plus and minus infinity, with
parameters cannot be computed efficiently, the implicit knowledge that all points outside
then a real time system cannot be realized of the current frame are zero (thus making
using the technique. For any LPC system, the sums tractable).
the basic LPC parameter determination is N  ∞
1  1
∞
always the same. We are faced with the ∑ a i⋅ 
M ∑s ( k − i) s ( k − j)
M
⋅ ∑ s ( k) s ( k − j)
need to minimize the squared prediction i=1 k = − ∞  k =−∞
error, defined as
Substituting m for k – l,
2
N
1 ∞  1
∞

1
M  N  ∑ a i⋅ 
M ∑ s ( m) s ( m + j − i)
M
⋅ ∑ s ( m) s ( m + j)

ε
M
⋅ ∑  s ( k) −
 ∑ a i⋅ s ( k − i) i=1 k = − ∞  k =−∞

k =1  i=1  which, because of the limits on the sums

where N is the model order, M the length of and because
the speech segment under consideration, ai
∞
is the ith LP coefficient, and s(k) is the kth 1
speech sample within the analysis frame. R( j)
M
⋅ ∑ s ( m) s ( m + j )
Differentiating this equation with respect to m=−∞
the a’s, then setting equal to zero, we come
to we can write in matrix form

RA=C
2 The fundamental difference between these
Digital Compression for Multimedia, Jerry two representations is the form of the
D. Gibson et. al.
covariance or correlation matrix. In the
autocorrelation system, the R matrix is
Toeplitz. In the covariance representation,
this is not the case. Toeplitz matrices lend
themselves to a particularly elegant,
efficient, and recursive solution known as
the Durbin or Levinson-Durbin algorithm.
Durbin’s recursion computes the LP
coefficients one at a time in terms of the
previous coefficient. While the covariance
coefficients can be calculated somewhat
more efficiently than brute force matrix
inversion through Cholesky decomposition,
the Levinson-Durbin algorithm has a decided
important and single most difficult, part of
the LPC vocoder process.
On the surface, the determination
problem would not seem to be all that
difficult. A quick survey of many of the
characteristics of voiced speech versus
unvoiced speech would lead one to expect
that the two are easily separable. Let’s take
such a survey. First, lets compare the
energy content of voiced and unvoiced
speech segments.
3000
2000

computational advantage. Additionally, the 1000

filter coefficients resulting from the 0

Levinson-Durbin algorithm produce an -1000

inherently stable inverse filter3. Both -2000

0 1000 2000 3000 4000 5000 6000
methods find widespread use, though the Log Energy
literature seems to favor the autocorrelation 60

approach (while it should be noted that the 50

Federal LPC-10 standard is based on the 40
covariance method). One other method of
note is the lattice based LP filter, which is 30

beyond this presentation, but uses a lattice 20

0 10 20 30 40 50 60
based inverse synthesis filter approach. For
comparison, see Table 1 – Storage and Figure 5 - Log energy of the utterance of the word
"six"
computational burden of LPC methods for
an analysis of the three methods4
The central portion of figure 5 shows the
Table 1 – Storage and computational vowel (voiced) portion of the utterance “six”
burden of LPC methods as spoken by a male speaker. As the graph
shows, the energy content of the voiced
Storage Covariance Auto- Lattice portion of the signal is on the order of 15-
correlation 20dB higher than that of the unvoiced
Data M M 3M
portion, and 40dB higher than those areas
Matrix N2/2 N
of silence. This difference, which is typically
Window 0 M
maintained within a speaker for all speech,
Multiplies
Windowing 0 M
is one of the stronger indicators of speech
Correlation M*N M*N type.
Matrix N3 N2 5*M*N
soltn.

Speech type determination

Aside from calculation of the LP
coefficients, determination of the speech
type for a given frame is the single most

3
Digital Processing of Speech Signals, L.R.
Rabiner, R. W. Schafer, pg. 418.
4
Ibid
 Another metric widely used in
speech type analysis is the zero crossing
3000
2000
1000
0
-1000
-2000
0 1000 2000 3000
Zero-crossing rate
0.8
0.6
0.4
0.2
0
0 10 20 30
Figure 6 - Zero crossing rate, utterance of the word
“six”
rate. For voiced speech, the majority of the
speech energy is concentrated well below
3kHz. For unvoiced speech, considerable
energy content is still present at and above
3kHz. Since low frequencies imply low zero
crossing rates, and high frequencies imply
high zero crossing rates, the zero cross rate
normalized normalized to the block length is
highly correlated with the energy content of
the speech segment, which is in turn highly
correlated with the speech type (see figure
6). Note how the zero crossing rate is very
high for the ‘s’ and last portion of the ‘x’
sound, is much lower for the vowel portion,
and is almost zero for the silent periods
(notice that the word “six” has a stop in its
production, where the excitation source to
the vocal tract is temporarily blocked,
producing momentary silence).
Zero crossing rate is very sensitive
to any DC offset in the signal, which
necessitates a high pass filter prior to its
determination in signals of unknown
recording origin. It is also sensitive to
broadband background noise, which
becomes a much harder problem to
eliminate in practice.
As stated earlier, in some vocoders
the V/U/S determination is made after the
calculation of the LP coefficients. This is
because there is considerable information
about the nature of the speech sample
embedded in those coefficients. For voiced
speech, with its characteristically lower
frequency content, there is usually a strong
correlation from one sample to the next.
Unvoiced speech, being much closer to
random, does not have this property, and
the difference manifests itself in properties
of the LP coefficients, most notably the first
LP coefficient. Consider figure 7, which
4000 5000 6000 shows the first LP coefficient calculated for
segments of the utterance of “six” we’ve
been using. Notice how the first coefficient
clusters around the value 1 for the unvoiced
3000
2000
40 50 60 1000
-1000
-2000
0 1000 2000 3000 4000 5000 6000
First prediction coefficient
2
1
0
-1
-2
0 10 20 30 40 50 60
Figure 7 - First LP coefficient, utterance of the
word “six”
portions of the speech, but is clustered
around -1 for the voiced section and for the
silence. This shows some specificity in
determining voiced from unvoiced speech,
but little ability to distinguish between
voiced speech and silence. Other
difference metrics exist, and the topic is still
under consideration 30 years after speech
coding first became a hot research topic.
Consider figures 8, 9, and 10, which show a
set of indicators used in a speech
classification system by Atal and Rabiner5.
Shown are histograms for the distribution of
five metrics of speech type, including zero
crossing rate, first autocorrelation coefficient
measurement, log energy, first LPC
5
A Pattern Recognition Approach to Voiced-
Unvoiced-Silence classification with
Applications to Speech Recognition, Bishnu
S. Atal, Lawrence R. Rabiner, IEEE
Transactions on Acoustics, Speech, and
Signal Processing, June 1976
coefficient, and LPC estimation error from a
variety of speakers and speech samples.
What is obvious in these distributions is that
significant differences exist between voiced
speech, unvoiced speech, and silence which
should be able to discriminate between the
three. Equally obvious, however, is the
overlap in the distributions that render any
one of the metrics, taken by itself, as
insufficient to the task. Some manner of
maximum likelihood estimator must be
employed which minimizes the probability of
making a false determination. Once useful
solution to the problem assumes a
multidimensional Gaussian distribution with
known mean mi and covariance matrix Wi,
where i = 1,2,3 corresponds to silence,
unvoiced, or voiced speech. The L-
dimensional Gaussian density function for
such a system has, given a measurement x
which is a column vector of distance
metrics,
Our decision rule which minimizes
classification errors states that the
measurement vector x should be assigned
to class i if
where pi is the a priori probability that x
belongs to class i.
This expression for the decision
metric can be simplified, since it is a
monotonically increasing function, as
Once all three distance measures (for
silence, unvoiced, and voiced decisions) are
calculated, classical Bayesian probability
provides the joint probabilities as
The highest P is chosen as the speech type.
The only remaining part of the
speech decision metric is the calculation of
the covariance matrices between the five
metrics. This constitutes the learning
portion of the algorithm. In my project, I
did not have the time to perform extensive
learning on speech samples. Instead, I
attempted to use the values for the W
matrices as provided by Rabiner and Atal,
computed for various sets of training data.
This implies that the decision making portion
of my vocoder was based on assumptions
that did not hold directly for the conditions
in which my speech was recorded. As a
practical point, their algorithm assumed a
specific scaling of data and block size that I
had to emulate because they did not use a
normalized metric for the zero crossing rate.
Ultimately, this technique failed to
adequately determine speech type, so I fell
back on a simpler method employing the
first LPC coefficient, the zero crossing rate,
and the signal energy optimized for the
speech signals with which I was working.

Since the last two terms don’t depend on x,

Atal and Rabiner claim that they only
introduce a negligent bias towards the ith
class, and don’t provide any “significant
advantage” over simplifying the decision
rule to only use the measurement vector x,
which thus reduces our distance metric to
be

Figures 8 & 8 - Speech type estimators
So why is the speech type
determination so important? Recall that the
speech synthesizer uses the speech type to
choose the excitation source for each frame.
Failure to chose the right type of excitation
produces speech that, while intelligible,
sounds very wrong to the human ear.
Voiced speech excited with a noise source
tends to sound “breathy”, or, as I call it,
Borg-like (an example is included on the
sample slide of the power point
presentation.). Unvoiced speech that is
excited with a periodic input sounds
synthetic or “robotic”, something like one
hears from a child’s speech synthesizer.
Occasional mistakes in the classification
cause individual frames to sound wrong in
the synthesized speech – an example of a
fully processed sentence is included in the
presentation that has a “chirp” where
unvoiced speech was mistaken for voiced
speech, a high pitch rate detected, and the
resultant frame stands out clearly to the ear
as in error.
Pitch detection
The final important step in LPC speech
processing is the detection of pitch in voiced
speech segments. During normal human
speech production, we vary the frequency of
the glottal pulse to produce the normal ups
and downs of the primary frequency content
for that speech, much as we do when
singing, though not nearly as pronounced.
While we are perceptually largely unaware
of this in normal conversation, a speech
synthesizer that does not preserve this part
of the speech signal will sound monotonic –
perhaps you’ve heard Steven Hawking’s
speech coder that produces all speech at a
single excitation frequency (again, an
Figure 10 - Speech type estimators cont'd
example is provided in the presentation).
Theoretically, pitch detection is
straightforward. A Fourier transform of the
frame can typically pick out the fundamental
frequency for any given frame. The
problem, of course, is the processing power
7
x 10
14
12
10
2
0
-2
-4
0 50 100 150 200 250
Figure 9 - Autocorrelation of voiced speech
required to make this determination. In
principle, the autocorrelation function for a
frame ought to work equally well – samples
that are related by the fundamental period
or its harmonics in the frame should show
the highest correlation. In practice, this
often works, but can easily be confounded
by the specifics of the vocal tract response
that may shape the envelope of the speech
signal in such a was as to accentuate the
harmonics of the fundamental period,
creating peaks that may actually exceed
the fundamental. Additionally, computing
a true autocorrelation sequence is an
expensive operation, requiring many
multiplies.
In fact, the information content of
a full autocorrelation is excessive to the
rather simpler task of pitch detection.
Small excursions and noise in the signal
ought not contribute anything to the pitch
detection process. Therefore, several
methods can be found that use a modified
form of input signal as input for pitch
detection. Once such method, proposed
by Sondhi6, is to use a highly non-linear
transformation called center clipping.
In center clipped speech, a variable clipping
level is determined, usually as some fixed
Y(n)=C[x(n)]
-CL
+CL
Figure 11 - Center clipping
percentage of the maximum signal
amplitude within the frame. All signal
300
samples falling below this threshold are set
to 0, and all values above the value in
absolute value retain that value. This
“compresses” the center out of the signal,
leaving only the excursions above and below
the clipping threshold. The modified speech
6
“New Methods of Pitch Extraction”, M.M.
Sondhi, IEEE Trans. Audio and
Electroacoustics, June 1968
can then be autocorrelated and a peak
detector employed to locate the
fundamental.
For many applications, especially
low lag voice coders, the autocorrelation is
still too time consuming and unnecessarily
complex. To reduce this complexity, the
Average Magnitude Difference Function, or
AMDF7. The AMDF is defined as
γ n (k ) = ∑m =−∞ x(n + m) − x(n + m − k )
∞
technique has been used with some
success.
In all of these algorithms, pitch
period estimates contain errors where a
harmonic was mistaken for a fundamental,
or where the pitch was simply miscalculated.
It has become almost universal to follow the
pitch detector with a 5th order median filter.
A median filter does not behave like an
averaging filter, which has the form of a low

pass filter. Median filters do a better job of

throwing out outliers in the pitch estimation
For a periodic sequence, x(n) – x(n-k) while retaining the shape of the pitch
should be zero for k = 0, P, 2P, etc, where contour.
the difference between sample values goes
to zero at multiples of the fundamental
frequency. Thus, the output of the AMDF is Shortcomings and Fixes
a function that shows distinct dips at the
So where does the LPC model break
fundamental period. A suitable negative
down? The most obvious problem with LPC
peak detector may then be employed to
in general is that human speech production,
extract the pitch period.
while sufficiently modeled by an all pole
In practice, the AMDF does not add
inverse filter, is not strictly an all pole
any processing capability to a pitch detector,
system. Several zeros may be present in
and is no better than the autocorrelation for
speech – these come from radiation effects
that purpose. Its main benefit is in its
at the lips, nasals (the sounds like ‘m’ and
computational simplicity, which involves no
‘n’ that are produced through the nasal
multiplies, and hence has found wide
cavity with the mouth mostly closed), and
acceptance in low lag speech coding
from the glottal pulse itself. There is no
applications.
simple fix for this problem in an LPC vocoder
A more robust, and commensurately
unless one is willing to model the system
more complicated pitch detector goes by the
with an IIR structure. The success of
acronym SIFT, for Simple Inverse Filtering
commercial vocoders that do not require this
Tracking. In the SIFT algorithm, the speech
complexity, and which produce reasonable
is low-pass filtered to 900Hz, then
voice quality at low bit rates, are testament
decimated 5:1. This produces a 1 kHz wide
to how well the all pole model actually
smaller signal for analysis which, because of
works in general.
the human speech process, still retains all of
One widely used method to account
the important formant information. That
for radiation effects is pre-emphasis. This
decimated signal is then LPC analyzed with
spectrally flattens the signal to ameliorate
a 4th order autocorrelation model filter. The
the roll-off at high frequencies. This roll-off
decimated input is inverse filtered using this
demands that the LP coefficients retain high
4th order filter, producing the prediction
precision, in direct competition with low bit
error signal, which is approximately
rates. Typically, the speech signal is passed
spectrally flat. Its autocorrelation is taken,
through a filter of the form
and a peak picker extracts the fundamental
frequency, which is then interpolated to
improve its estimation accuracy. Citations in H ( z ) = 1 − αz −1
the literature seem to suggest that this
that has a characteristic high pass
7
quality but is computationally quite simple to
“Average Magnitude Difference Function implement. Alpha in the range of .9 to .95
Pitch Extractor”, M.J.Ross, H.L.Shaffer, et. is typical. To reverse the effects of the pre-
Al, IEEE Trans. Acoust. Speech and Signal emphasis, the synthesized speech signal
Proc, October 1974.
must be passed through a corresponding
inverse de-emphasis filter. This approach
was employed in my simple Matlab based
vocoder.
As a practical matter, stability
concerns may sometimes plague the inverse
synthesis filter when too few bits are used
at some point in the analysis or transmission
of the LPC model. Even though the
autocorrelation method is guaranteed to
produce a stable filter in theory, this is only
guaranteed when coefficient round-off is not
an issue. Any loss of precision in coefficient
generation can push a pole outside of the
unit circle and cause the speech synthesis to
explode at the other end. A practical fix for
this problem in low bit rate coders is what
are known as PARCOR coefficients, or
PARtial CORrelation coefficients. These
coefficients model the reflections that occur
at the impedance mismatched tubes in the
vocal tract model. They are limited in
magnitude to be less than 1, and are easily
calculated using a variation of the Levinson-
Durbin recursion, and just as easily inverted
into filter tap weights at the receiver. These
coefficients also provide a robust stability
check on any speech frame – PARCOR
Amplitude 0.015-.03 second
normalization Frame segmentation
Zero crossing rate detection
Gain estimation Log energy detection
Prediction error analysis
Pitch period impulse
train generator
Excitation selection
Random noise
generator
Figure 12 - Vocoder realization
coefficients exceeding 1 in absolute value
indicate that instability has crept into the
calculation, and an intelligent processor can
deal with the problem.
Another technique used to improve
the stability of the synthesis filter is called
bandwidth expansion. The filter coefficients
are modified by the following equation:
a new = γ i ⋅ ai
where gamma is a positive constant less
than 1 (typical values range from 0.988 to
0.996)8. The effect of this adjustment is to
broaden the spectrum of the speech signal,
especially around the formant peaks. This
also shortens the impulse response of the
inverse filter, aiding in stability.
Other issues with the LPC vocoder
are based on its block structure processing
of speech. Transition frames are frames of
speech that are moving from one type of
speech to another. Traditional LPC cannot
model this problem, and so is forced to
make a binary decision. Pitch synchronous
LPC attempts to solve this problem by
fracturing the speech signal not on fixed
temporal boundaries, but rather on natural
Order 12 LPC
Pre-emphasis Hanning
Parameter
1-0.9z window
estimation
Center clipped
V/U/S 5 sample pitch
Correlation pitch
decision median filter
extraction
De-emphasis
Inverse LPC filter
1/1-.9z
8
“Speech Coding Algorithms”, Wai C. Chu,
pg. 133
speech boundaries where each frame
contains only a single type of speech. I
have no working experience with this type
of encoder, and cannot attest to its worth,
but it has apparently found its way into
commercial products.
The binary speech type decision
also causes problems for LPC. Certain kinds
of sounds in speech do not fall cleanly into
either the voiced or unvoiced category –
these include sounds like the letter ‘z’ and
some of the nasals. These sounds show a
noticeable mix of excitations, appearing as
noisy periodic signals. Handling this
problem in LPC has created a class of mixed
excitation coders that use both periodic and
noise components, suitably balanced, as the
excitation source for the synthesizer.

Finally, I would like to mention the vocoder

that I created and tweaked to perform the
experiments and to generate the sample
waveforms for this presentation. Refer to
Figure 12 - Vocoder realization for a block
diagram. As usual, what is theoretically a
simple task becomes a devil-in-the-details
engineering problem. By far, the V/U/S
decision block was the most difficult task, as
the literature had quite properly warned me.
Nonetheless, the project provided many
hours of pain and pleasure, and I quite
enjoyed the opportunity to build and to
present it.