Field Of The Invention
[0001] The present invention relates to speech encoding and decoding systems in general,
and more particularly to systems and methods for vector quantization of line spectral
frequencies.
Background Of The Invention
[0002] One of the challenges of designing vector quantizers for linear predictive coding
(LPC) speech filters is making them robust to variations in spectral balance, as well
as variations between speakers. Spectral balance variations may have several sources,
but the dominant sources are the spectral responses of microphones and of anti-aliasing
filters, which can vary quite considerably. In order to account for these variations
it is common to train a quantizer for an LPC filter for use with a wide variety of
speech input, recorded from many different sources.
[0003] Conventional vector quantization (VQ) training methods use speech from as many different
sources as possible, in an attempt to provide robust performance for many different
input spectra. However, this approach is disadvantageous in that training is relatively
slow and complex, as many speech samples are required. This approach furthermore generally
results in a quantizer which is not optimal for any one filtering condition.
[0004] One method for handling spectral balance variations is the Microphone/Speaker Adaption
(MSA) method taught by Aarskog
et al. In this method, the average spectrum of speech input presented to a speech coder
is compensated for by an MSA filter prior to further compression by an inverse filter.
The speech is subsequently filtered by a complementary filter after decoding. This
method is disadvantageous in that it requires two stages of inverse filtering, thus
increasing the complexity of the quantizer due to the required autocorrelation function
calculations. Furthermore, two LPC filter quantizers are needed, one for the MSA filter
and one for the conventional LPC filter. An additional slow-speed data path is also
needed to convey the quantized MSA filter parameters from encoder to decoder.
[0005] The following publications are believed to be descriptive of the current state of
the art of speech encoding and decoding systems in general, and vector quantization
of line spectral frequencies technologies in particular, and terms related thereto:
A. Aarskog, A. Nilsen, O. Berg, and H. C. Gruen, "A Long-Term Predictive ADPCM Coder
with Short-Term Prediction and Vector Quantization," 1991 International Conference
on Acoustics, Speech, and Signal Processing, ICASSP-91, vol. 1, pp. 37 -40;
A. Aarskog and H. C. Gruen, "Predictive Coding of Speech Using Microphone/Speaker
Adaptation and Vector Quantization," IEEE Transactions on Speech and Audio Processing,
April 1994, vol. 22, pp. 266 -273;
W.B. Kleijn and K.K. Paliwal, "Speech Coding and Synthesis," Elsevier Press, 1995.
[0006] The disclosures of all patents, patent applications, and other publications mentioned
in this specification are hereby incorporated by reference.
Summary Of The Invention
[0007] The present invention seeks to provide improved systems and methods for vector quantization
that account for spectral balance variations while avoiding the limitations of the
prior art.
[0008] A quantization system and method are disclosed that achieve similar objective performance,
in terms of mean spectral distortion and outliers, for speech within and outside the
training database, and similar quantizer performance for different types of speech
largely irrespective of the spectral balance. The present invention exploits properties
of line-spectrum pairs to yield a robust quantizer with superior performance under
various conditions. The present invention further discloses a more error-robust system
and method for deriving adaptable mean values based upon previous quantizer decisions
in a uniform gain moving average fashion.
[0009] The present invention is an extension to mean-removed vector quantization and is
equally applicable to both auto-regressive and moving average predictive vector quantization.
A system and method are disclosed for slow averaging of the positions of the inverse
quantized line spectral frequencies (LSFs) using a series of simple filters (one per
LSF) with one or more long time constants.
[0010] There is thus provided in accordance with a preferred embodiment of the present invention
a method of providing robust quantization of speech spectral parameters tolerant to
spectral balance and speaker variations, the method including the steps of, for each
of a plurality of line spectral frequencies (LSFs) of a speech spectrum, quantizing
the displacement of the LSF from an estimate of its long-term mean, reconstructing
an estimate of the LSF from the quantized displacement and the long-term LSF mean
estimate, and filtering the reconstructed LSF estimate, thereby providing a subsequent
long-term LSF mean estimate.
[0011] Further in accordance with a preferred embodiment of the present invention the filtering
step includes filtering the reconstructed LSF estimate using a first-order recursive
filter.
[0012] Still further in accordance with a preferred embodiment of the present invention,
the first-order recursive filter is of unity gain and employs a time constant of about
1 second for the LSF.
[0013] There is also provided in accordance with a preferred embodiment of the present invention
a method of quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, the method including the steps of, for each of a plurality
of line spectral frequencies (LSFs) of a speech spectrum, at an encoder a) quantizing
the difference between the LSF and a current LSF mean value estimate, and at the encoder
and a decoder b) dequantizing the difference, c) adding the dequantized difference
to a current LSF mean value estimate, thereby providing an approximation of the LSF,
and d) filtering the quantized LSF together with the current LSF mean value estimate,
thereby providing a new current LSF mean value estimate.
[0014] There is additionally provided in accordance with a preferred embodiment of the present
invention a method of quantizing speech spectral parameters that is tolerant to spectral
balance and speaker variations, the method including the steps of, for each of a plurality
of line spectral frequencies (LSFs) of a speech spectrum, at an encoder a) quantizing
a prediction error derived from the LSF from which a current short-term LSF mean value
and a current moving average predicted LSF estimate have been subtracted, and at the
encoder and a decoder b) dequantizing the prediction error, c) determining a next-current
short-term LSF mean value from the dequantized prediction error and at least one previously
dequantized prediction error, and d) determining a next-current moving average predicted
LSF estimate from the dequantized prediction error and at least one previously dequantized
prediction error.
[0015] Further in accordance with a preferred embodiment of the present invention the next-current
short-term LSF mean value is the sum of a training data derived mean and a moving
average of a plurality of previously dequantized prediction error values.
[0016] Still further in accordance with a preferred embodiment of the present invention
the equal gains are assigned to each dequantized prediction error value.
[0017] There is also provided in accordance with a preferred embodiment of the present invention
apparatus for providing robust quantization of speech spectral parameters tolerant
to spectral balance and speaker variations, the apparatus including means for quantizing
the displacement of a line spectral frequency (LSF) from an estimate of its long-term
mean, means for reconstructing an estimate of the LSF from the quantized displacement
and the long-term LSF mean estimate, and means for filtering the reconstructed LSF
estimate, thereby providing a subsequent long-term LSF mean estimate.
[0018] Further in accordance with a preferred embodiment of the present invention the filtering
means includes a first-order recursive filter.
[0019] Still further in accordance with a preferred embodiment of the present invention
the first-order recursive filter is of unity gain and employs a time constant of about
1 second for the LSF.
[0020] There is additionally provided in accordance with a preferred embodiment of the present
invention apparatus for quantizing speech spectral parameters that is tolerant to
spectral balance and speaker variations, the apparatus including an encoder including
means for quantizing the difference between a line spectral frequency (LSF) and a
current LSF mean value estimate, means for dequantizing the difference, means for
adding the dequantized difference to a current LSF mean value estimate, thereby providing
an approximation of the LSF, and means for filtering the quantized LSF together with
the current LSF mean value estimate, thereby providing a new current LSF mean value
estimate, and a decoder including means for dequantizing the difference, means for
adding the dequantized difference to a current LSF mean value estimate, thereby providing
an approximation of the LSF, and means for filtering the quantized LSF together with
the current LSF mean value estimate, thereby providing a new current LSF mean value
estimate.
[0021] There is also provided in accordance with a preferred embodiment of the present invention
apparatus for quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, the apparatus including an encoder including means for quantizing
a prediction error derived from the LSF from which a current short-term LSF mean value
and a current moving average predicted LSF estimate have been subtracted, means for
dequantizing the prediction error, means for determining a next-current short-term
LSF mean value from the dequantized prediction error and at least one previously dequantized
prediction error, and means for determining a next-current moving average predicted
LSF estimate from the dequantized prediction error and at least one previously dequantized
prediction error and the current short-term LSF mean value, and a decoder including
means for dequantizing the prediction error, means for determining a next-current
short-term LSF mean value from the dequantized prediction error and at least one previously
dequantized prediction error, and means for determining a next-current moving average
predicted LSF estimate from the dequantized prediction error and at least one previously
dequantized prediction error.
[0022] Further in accordance with a preferred embodiment of the present invention the next-current
short-term LSF mean value is the sum of a training data derived mean and a moving
average of a plurality of previously dequantized prediction error values.
[0023] Still further in accordance with a preferred embodiment of the present invention
the equal gains are assigned to each dequantized prediction error value.
Brief Description Of The Drawings
[0024] The present invention will be understood and appreciated more fully from the following
detailed description taken in conjunction with the appended drawings in which:
Fig. 1 is a simplified illustration of a system for backwards-adaptive vector quantization
of line spectral frequencies (LSF), constructed and operative in accordance with a
preferred embodiment of the present invention;
Fig. 2 is a simplified illustration of a system for backwards-adaptive vector quantization
of line spectral frequencies (LSF), constructed and operative in accordance with another
preferred embodiment of the present invention; and
Figure 3 is a simplified graph illustration showing mean spectral distortion performance
(dB) of the systems of Figs. 1 and 2 with fixed means, moving average mean adaptation
and backwards adapted means.
Detailed Description Of Preferred Embodiments
[0025] Reference is now made to Fig. 1, which is a simplified illustration of a system for
backwards-adaptive vector quantization of line spectral frequencies (LSF), constructed
and operative in accordance with a preferred embodiment of the present invention.
In the system of Fig. 1 LSFs are quantized with their previous long-term mean values
removed, using any conventional VQ technique, such as memoryless, AR predictive, MA
predictive, or other suitable technique. The same long-term mean value is used during
encoding and decoding. As each quantization process is performed, the long-term average
value of the LSF changes at both the encoder and decoder. In this way, the quantizer
adapts to long-term variations in the LSFs.
[0026] In the system of Fig. 1 line spectral frequencies of a speech spectrum are provided
to an encoder, generally referenced 10. A subtractor 12 subtracts the current estimate
of the mean value associated with the LSF from the LSF input. A quantizer 14 then
quantizes the difference of the LSF from its mean value by selecting an appropriate
codebook index in accordance with any known and suitable quantization means. The quantization
index is then provided to inverse quantizers 16 and 18 at encoder 10 and a decoder,
generally referenced 20, respectively. Inverse quantizer 18 dequantizes the quantization
index using any known and suitable means to determine an associated LSF. An adder
22 adds the current estimate of the mean value associated with the LSF back into the
LSF determined at inverse quantizer 18, thus providing an approximation of the LSF
input to encoder 10.
[0027] The quantized LSF from adder 22, in addition to being used during subsequent speech
encoding and decoding, is provided to a simple, first-order filter where the LSF is
multiplied by a filter value X at a multiplier 26. The previous estimate of the LSF
mean value, held at a delay 28, is then multiplied by a filter value 1-X at a multiplier
30. The result of multiplier 30 is then added to the result from multiplier 26 at
an adder 32. The result from adder 32 represents the current estimate of the LSF mean
value and is stored in delay 28.
[0028] The time constant used in the system of Fig. 1 may in principle take any value, however,
the filter value X is preferably determined such that the time constant of the filter
is relatively long compared to the maximum duration of steady state vowels. This ensures
that the filter removes the slow-varying spectral shape rather than the utterance-to-utterance
variations, i.e., the fast spectral variations of normal speech, which typically do
not exceed a few hundred milliseconds. Too long a time constant restricts the time
needed to adapt to new speakers. Experimentation has shown that a time constant of
approximately 1 second, corresponding to a filter value of X=.037, provides satisfactory
performance. Where errors may occur in quantizer index transmission, the time constant
is preferably selected to minimize error propagation stemming from the use of an infinite
memory recursive filter.
[0029] Inverse quantizer 16 likewise dequantizes the quantization index to determine an
associated LSF which is then provided to an adder 34 and a simple, first-order filter
which includes a multiplier 38, an adder 40, a delay 42, and a multiplier 44, all
of which operate in the manner described hereinabove for adder 22, multiplier 26,
delay 28, multiplier 30, and adder 32, with the notable exception that the estimate
of the LSF mean value in delay 42 is provided to subtractor 12 in addition to being
provided to adder 34.
[0030] Reference is now made to Fig. 2, which is a simplified illustration of a system for
backwards-adaptive vector quantization of line spectral frequencies (LSF), constructed
and operative in accordance with another preferred embodiment of the present invention.
In the system of Fig. 2 the LSF means are derived from a relatively long moving average
predictor in order to overcome the problems associated with infinite error propagation
and incorporated within a conventional third-order (short) moving average predictive
vector quantizer. The system of Fig. 2 may be implemented using a rectangular window
moving average predictor for the calculation of the LSF means, such as one that is
about 750 ms long (a relatively long predictor). This may be easily achieved by employing
a circular buffer containing the quantizer indices from previous decisions.
[0031] In the system of Fig. 2, line spectral frequencies of a speech spectrum are provided
to an encoder, generally referenced 50. A subtractor 52 receives the current short-term
mean value associated with the LSF from an adder 54 and subtracts it from the LSF.
A subtractor 56 then receives the current moving average predicted estimate of the
LSF from an adder 58 and subtracts it from the output of subtractor 52. The output
of subtractor 56 is then divided by a tap
t0 of the short MA predictor at a divider 92 to provide a prediction error which is
then quantized at a quantizer 60 using any known and suitable quantization means.
The quantization index is then provided to inverse quantizers 62 and 64 at encoder
50 and a decoder, generally referenced 66, respectively. Inverse quantizer 62 dequantizes
the quantization index using any known and suitable means to determine an associated
LSF. The taps of the short moving average predictor (
t0,
t1 &
t2) may be determined by any reasonable technique, but are ideally jointly optimized
with the relatively long moving average LSF mean predictor in operation.
[0032] The current output of inverse quantizer 62 is multiplied by
t0 at a multiplier 68 and provided to an adder 70. The previous output of inverse quantizer
62, stored at a delay 72, is multiplied by a tap
t1 at a multiplier 74 and provided to adder 70. The twice-previous output of inverse
quantizer 62, stored at a delay 76, is multiplied by a tap
t2 at a multiplier 78 and provided to adder 70. Adder 70 adds all three inputs and provides
the result to an adder 80. The output of adder 70 represents the current quantization
error component of the output LSF.
[0033] The previous output of inverse quantizer 62, stored at delay 72, is multiplied by
tap
t1 at a multiplier 82 and provided to adder 58. The twice-previous output of inverse
quantizer 62, stored at delay 76, is multiplied by tap
t2 at a multiplier 84 and provided to adder 58. Adder 58 adds the two inputs and provides
the result to subtractor 56. The output of adder 58 represents the current predicted
estimate of the LSF.
[0034] The current output of inverse quantizer 62 is also provided to an ordered series
of
n delays 86, with each delay storing an
nth previous output of inverse quantizer 62. Each previous value
n is then multiplied by 1/
n by a series of multipliers 88, thereby providing equal gain for each value
n, and provided to adder 54, where they are added together with a predetermined estimate
µ of the mean of the LSF stored at a delay 90. The value of µ may be determined from
training data and represents an initial estimate of the LSF means. The output of adder
54 is then provided to adder 80 as well as subtractor 52. The output of adder 54 represents
the current short-term mean value associated with the LSF.
[0035] The current quantization error component of the LSF is preferably added to the current
short-term LSF mean value at adder 80 to provide an approximation of the LSF input.
[0036] At decoder 66, elements 68' - 90' preferably operate in the manner described hereinabove
for correspondingly-numbered elements 68 - 90 with the notable exceptions that adder
54' provides input only to adder 80', delay 72' provides input only to multiplier
74', and delay 76' provides input only to multiplier 78'.
[0037] Experimentation with the system of Fig. 2 has shown that there is little degradation
in performance when the moving-average derived adaptive mean method is applied to
a third-order moving average predictive VQ as compared with a conventional third-order
MA predictive VQ without mean adaption.
[0038] Experimentation has shown that the application of the systems of Figs. 1 and 2 leads
to significant gains in performance since the long-term averaging of the LSF means
removes some of the speaker and microphone/anti-aliasing spectral variation which
is present in the input. Such performance gains are shown in Fig. 3 which is a simplified
graph illustration showing Mean Spectral Distortion Performance (dB) of a conventional
third-order MA Predictive LSF VQ with Fixed Means, represented by a plot 100, the
Moving Average Mean Adaptation of Fig. 2, represented by a plot 102, and the Backwards
Adapted Means of Fig. 1, represented by a plot 104. Fig. 3 shows the spectral distortion
figures for three identical third-order moving average predictive quantizers (MA-PVQs)
plotted with and without adaptation of the mean values as described hereinabove. The
test file that was used comprised 8,000 frames each of flat filtered speech, Intermediate
Reference System (IRS) filtered speech, and modified IRS filtered speech. The training
data that was used for both quantizers was IRS filtered.
[0039] While the methods and apparatus disclosed herein may or may not have been described
with reference to specific hardware or software, the methods and apparatus have been
described in a manner sufficient to enable persons of ordinary skill in the art to
readily adapt commercially available hardware and software as may be needed to reduce
any of the embodiments of the present invention to practice without undue experimentation
and using conventional techniques.
[0040] While the present invention has been described with reference to a few specific embodiments,
the description is intended to be illustrative of the invention as a whole and is
not to be construed as limiting the invention to the embodiments shown. It is appreciated
that various modifications may occur to those skilled in the art that, while not specifically
shown herein, are nevertheless within the true scope of the invention.
1. A method of providing robust quantization of speech spectral parameters tolerant to
spectral balance and speaker variations, the method comprising the steps of, for each
of a plurality of line spectral frequencies (LSFs) of a speech spectrum:
quantizing (14) the displacement (12) of said LSF from an estimate (28) of its long-term
mean;
reconstructing (22) an estimate of said LSF from said quantized displacement and said
long-term LSF mean estimate; and
filtering (26,30,32) said reconstructed LSF estimate, thereby providing a subsequent
long-term LSF mean estimate (28).
2. A method according to claim 1 wherein said filtering step comprises filtering said
reconstructed LSF estimate using a first-order recursive filter (26,30,32).
3. A method according to claim 2 wherein said first-order recursive filter is of unity
gain and employs a time constant of about 1 second for said LSF.
4. A method of quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, the method comprising the steps of, for each of a plurality
of line spectral frequencies (LSFs) of a speech spectrum:
at an encoder (10):
a) quantizing (14) the difference (12) between said LSF and a current LSF mean value
estimate;
at said encoder (10) and a decoder (20):
b) dequantizing (16,18) said difference;
c) adding (22,34) said dequantized difference to a current LSF mean value estimate
(28,42), thereby providing an approximation of said LSF; and
d) filtering (26,30,32;38,40,44) said quantized LSF together with said current LSF
mean value estimate (28,42), thereby providing a new current LSF mean value estimate.
5. A method of quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, the method comprising the steps of, for each of a plurality
of line spectral frequencies (LSFs) of a speech spectrum:
at an encoder (50):
a) quantizing (60) a prediction error (92) derived from said LSF from which a current
short-term LSF mean value (52) and a current moving average predicted LSF estimate
(56) have been subtracted; and
at said encoder (50) and a decoder (66):
b) dequantizing (62,64) said prediction error;
c) determining a next-current short-term LSF mean value (54) from said dequantized
prediction error and at least one previously dequantized prediction error (86); and
d) determining a next-current moving average predicted LSF estimate (58) from said
dequantized prediction error (72) and at least one previously dequantized prediction
error (76).
6. A method according to claim 5 wherein the next-current short-term LSF mean value (54)
is the sum of a training data derived mean (90) and a moving average (88) of a plurality
of previously dequantized prediction error values (86).
7. A method according to claim 6 wherein equal gains (88) are assigned to each dequantized
prediction error value (86).
8. Apparatus for providing robust quantization of speech spectral parameters tolerant
to spectral balance and speaker variations, said apparatus comprising:
means (14) for quantizing the displacement (12) of a line spectral frequency (LSF)
from an estimate (28) of its long-term mean;
means (22) for reconstructing an estimate of said LSF from said quantized displacement
and said long-term LSF mean estimate; and
means (26,30,32) for filtering said reconstructed LSF estimate, thereby providing
a subsequent long-term LSF mean estimate (28).
9. Apparatus according to claim 8 wherein said filtering means comprises a first-order
recursive filter (26,30,32).
10. Apparatus according to claim 9 wherein said first-order recursive filter is of unity
gain and employs a time constant of about 1 second for said LSF.
11. Apparatus for quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, said apparatus comprising:
an encoder (10) comprising:
means (14) for quantizing the difference (12) between a line spectral frequency (LSF)
and a current LSF mean value estimate;
means (16) for dequantizing said difference;
means (34) for adding said dequantized difference to a current LSF mean value estimate
(42), thereby providing an approximation of said LSF; and
means (38,40,44) for filtering said quantized LSF together with said current LSF mean
value estimate (42), thereby providing a new current LSF mean value estimate; and
a decoder comprising:
means (18) for dequantizing said difference;
means (22) for adding said dequantized difference to a current LSF mean value estimate
(28), thereby providing an approximation of said LSF; and
means (26,30,32) for filtering said quantized LSF together with said current LSF mean
value estimate (28), thereby providing a new current LSF mean value estimate.
12. Apparatus for quantizing speech spectral parameters that is tolerant to spectral balance
and speaker variations, said apparatus comprising:
an encoder (50) comprising:
means (60) for quantizing a prediction error (92) derived from said LSF from which
a current short-term LSF mean value (52) and a current moving average predicted LSF
estimate (56) have been subtracted;
means (62) for dequantizing said prediction error;
means (54) for determining a next-current short-term LSF mean value from said dequantized
prediction error and at least one previously dequantized prediction error (86); and
means (58) for determining a next-current moving average predicted LSF estimate from
said dequantized prediction error (72) and at least one previously dequantized prediction
error (76); and
a decoder (66) comprising:
means (64) for dequantizing said prediction error;
means (54') for determining a next-current short-term LSF mean value from said dequantized
prediction error and at least one previously dequantized prediction error (86'); and
means (58') for determining a next-current moving average predicted LSF estimate from
said dequantized prediction error (72') and at least one previously dequantized prediction
error (76').
13. Apparatus according to claim 12 wherein the next-current short-term LSF mean value
(54) is the sum of a training data derived mean (90) and a moving average (88) of
a plurality of previously dequantized prediction error values (86).
14. Apparatus according to claim 13 wherein equal gains (88) are assigned to each dequantized
prediction error value (86).