The present invention relates to the field of audio signals, more specifically to an approach
for processing an audio signal including a plurality of audio frames, wherein discontinuities
between consecutive filtered audio frames are reduced or omitted.
In the field of audio signal processing, an audio signal may be filtered for various reasons,
e.g., a long-term prediction filter may be used in an audio signal encoder, to attenuate or
even suppress completely a set of harmonics in the audio signal.
The audio signal includes a plurality of audio frames, and the frames are filtered using the
long-term prediction filter. When considering two consecutive frames of an audio signal, a
past frame and a current frame, a linear filter H(z) having a set of parameters c is used for
filtering the audio signal. More specifically, the past frame is filtered with the filter H(z)
using a first set of parameters c0 which will produce a so-called filtered past frame. The
current frame is filtered with the filter H(z) using a set of parameters C which will produce
a filtered current frame. Fig. 1 shows a block diagram for processing consecutive frames
of an audio signal in accordance with a known approach. An audio signal 100 including a
plurality of audio frames is provided. The audio signal 100 is supplied to a filter block 102
and a current frame n of the audio signal 100 is filtered. The filter block, besides the audio
signal 100, receives a set of filter parameters cn for the current frame of the audio signal.
The filter block 102 filters the current frame n of the audio signal and outputs a filtered
audio signal 104 including consecutive filtered frames. In Fig. 1, the filtered current frame
n, the filtered past frame n-1 and the filtered second last frame n-2 are schematically
depicted. The filtered frames are schematically represented in Fig. 1 with respective gaps
therebetween for schematically indicating a discontinuity 106a, 106b that may be
introduced by the filtering process between the filtered frames. The filter block 102 causes
filtering of the frames of the audio signal using respective filter parameters c0 and C for a
past frame n-1 and a current frame n. In general, the filter block 102 may be a linear filter
H(z), and one example for such a linear filter H(z) is the above mentioned long-term
prediction filter
H(z) = 1 - g - z
where the filter parameters are the gain "g" and the pitch lag "T". In a more general form,
the long-term prediction filter can be described as follows:
H(z) = 1 - g · A(z) · z
where A(z) is a FIR filter. A long-term prediction filter may be used to attenuate or even
suppress completely a set of harmonics in an audio signal. However, there is a high
probability of introducing a discontinuity 106a, 106b (see Fig. 1) between the filtered past
frame n-1 and the filtered current frame n when using such a long-term prediction filter
and when the past frame filter parameters c0 are different from the current frame filter
parameters . This discontinuity may produce an artifact in the filtered audio signal 104,
for example a "click".
Consequently, in view of the above described problems with the filtering of consecutive
frames resulting in discontinuities which, in turn, may produce undesired artifacts, a
technique is needed that removes a possible discontinuity. Several prior art approaches
dealing with the removal of a discontinuity of filtered frames of an audio signal are known
in the art.
In case the linear filter H(z) is a FIR filter, the current frame is filtered with the filter
parameters C of the current frame for producing a filtered current frame. In addition, a
beginning portion of the current frame is filtered with the filter parameters of the past
frame c0 for producing a filtered frame portion, and then an overlap-add or cross-fade
operation is performed over the beginning portion of the filtered current frame and the
filtered frame portion. Fig. 2 shows a block diagram of such a conventional approach for
processing consecutive audio frames for removing a discontinuity. When compared to Fig.
1, the filter block 102 includes a further processing block 108 for performing the overlapadd
or cross-fade operation. In the filtered audio signal 104, there will be no or a reduced
discontinuity between the consecutive filtered frames, as is schematically indicated in Fig.
2 showing the consecutive filtered frames n, n-1 and n-2 without the gaps of Fig. 1.
In other prior art approaches, the filter H(z) may be a filter having a recursive part, for
example an MR filter. In such a case, the approach as described above with regard to Fig.
2 is applied on a sample-by-sample basis. In a first step, the processing starts with the
first sample of the beginning portion of the current frame n being filtered with the filter
parameters c0 of the past frame n-1 yielding a first filtered sample. The sample is also
filtered with the filter parameters C of the current frame n producing a second filtered
sample. Then, the overlap-add or cross-fade operation is performed based on the first and
second filtered samples which yields the corresponding sample of the filtered current
frame n. Then the next sample is processed and the above steps are repeated until the
last sample of the beginning portion of the current frame n has been processed. The
remaining samples of the current frame n are filtered with the filter parameters C of the
current frame n.
Examples for the above mentioned known approaches for removing a discontinuity from
consecutive filtered frames are described, for example, in US 5,01 2,51 7 A in the context
of a transform coder, in EP 0732687 A2 in the context of a speech bandwidth expander, in
US 5,999,899 A in the context of a transform audio coder, or in US 7,353,1 68 B2 in the
context of a decoded speech postfilter.
While the above approaches are efficient for removing the undesired signal
discontinuities, since these approaches operate on a specific portion of the current frame,
the beginning portion, for being effective, the length of the frame portion has to be
sufficiently long, for example in the case of a frame length of 20 ms, the frame portion or
beginning portion length could be as long as 5 ms. In certain cases, this can be too long,
especially in situations where the past frame filter parameters c0 will not apply well to the
current frame and this may result in additional artifacts. One example is a harmonic audio
signal with fast changing pitch, and a long-term prediction filter that is designed to reduce
the amplitude of the harmonics. In that case, the pitch-lag is different from one frame to
the next. The long-term prediction filter with the pitch estimated in the current frame would
effectively reduce the amplitude of the harmonics in the current frame, but it would not
reduce the amplitude of the harmonics if used in another frame (e.g. beginning portion of
the next frame) where the pitch of the audio signal would be different. It could even make
things worse, by reducing the amplitude of non-harmonic-related components in the
signal, introducing a distortion in the signal
It is an object underlying the present invention to provide an improved approach for
removing discontinuities among filtered audio frames without producing any potential
distortion in the filtered audio signal.
This object is achieved by a method and an apparatus according to the independent
claims.
The present invention provides a method for processing an audio signal, the method
comprising removing a discontinuity between a filtered past frame and a filtered current
frame of the audio signal using linear predictive filtering.
The linear predictive filter can be defined as
with M the filter order and am the filter coefficients (with a = 1). This kind of filter is also
known as Linear Predictive Coding (LPC).
In accordance with embodiments, the method comprises filtering the current frame of the
audio signal and removing the discontinuity by modifying a beginning portion of the filtered
current frame by a signal obtained by linear predictive filtering a predefined signal with
initial states of the linear predictive filter defined on the basis of a last part of the past
frame.
In accordance with embodiments, the initial states of the linear predictive filter are defined
on the basis of a last part of the unfiltered past frame filtered using the set of filter
parameters for filtering the current frame.
In accordance with embodiments, the method comprises estimating the linear predictive
filter on the filtered or non-filtered audio signal.
In accordance with embodiments, estimating the linear predictive filter comprises
estimating the filter based on the past or current frame of the audio signal or based on the
past filtered frame of the audio signal using the Levinson-Durbin algorithm.
In accordance with embodiments, the linear predictive filter comprises a linear predictive
filter of an audio codec.
In accordance with embodiments, removing the discontinuity comprises processing the
beginning portion of the filtered current frame, wherein the beginning portion of the current
frame has a predefined number of samples being less or equal than the total number of
samples in the current frame, and wherein processing the beginning portion of the current
frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the
beginning portion of the filtered current frame.
In accordance with embodiments, the method comprises filtering the current frame of the
audio signal using a non-recursive filter, like a FIR filter, for producing the filtered current
frame.
In accordance with embodiments, the method comprises processing the unfiltered current
frame of the audio signal on a sample-by-sample basis using a recursive filter, like an MR
filter, and wherein processing a sample of the beginning portion of the current frame
comprises:
filtering the sample with the recursive filter using the filter parameters of the current frame
for producing a filtered sample, and
subtracting a corresponding ZIR sample from the filtered sample for producing the
corresponding sample of the filtered current frame.
In accordance with embodiments, filtering and subtracting are repeated until the last
sample in the beginning portion of the current frame is processed, and wherein the
method further comprises filtering the remaining samples in the current frame with the
recursive filter using the filter parameters of the current frame.
In accordance with embodiments, the method comprises generating the ZIR, wherein
generating the ZIR comprises:
filtering the M last samples of the unfiltered past frame with the filter and the filter
parameters used for filtering the current frame for producing a first portion of filtered
signal, wherein M is the linear predictive filter order,
subtracting from the first portion of filtered signal the M last samples of the filtered past
frame, filtered using the filter parameters of the past frame, for generating a second
portion of filtered signal, and
generating a ZIR of a linear predictive filter by filtering a frame of zero samples with the
linear predictive filter and initial states equal to the second portion of filtered signal.
In accordance with embodiments, the method comprises windowing the ZIR such that its
amplitude decreases faster to zero.
The present invention is based on the inventor's findings that the problems that have been
recognized in conventional approaches for removing signal discontinuities which result in
the additional unwanted distortion mentioned above, are mainly due to the processing of
the current frame or at least a portion thereof on the basis of the filter parameters for the
past frame. In accordance with the inventive approach this is avoided, i.e. the inventive
approach does not filter a portion of the current frame with the filter parameters of the past
frame and thus avoids the problems mentioned above. In accordance with embodiments,
for removing the discontinuity, an LPC filter (linear predictive filter) is used for removing
the discontinuity. The LPC filter may be estimated on the audio signal and therefore it is a
good model of the spectral shape of the audio signal so that, when using the LPC filter,
the spectral shape of the audio signal will mask the discontinuity. In an embodiment, the
LPC filter may be estimated on the basis of the non-filtered audio signal or on the basis of
an audio signal that has been filtered by a linear filter H(z) mentioned above. In
accordance with embodiments, the LPC filter may be estimated by using the audio signal,
for example the current frame and/or the past frame, and the Levinson-Durbin algorithm. It
may also be computed only on the basis of the past filtered frame signal using the
Levinson-Durbin algorithm.
In yet other embodiments, an audio codec for processing the audio signal may use a
linear filter H(z) and may also use an LPC filter, either quantized or not, for example to
shape the quantization noise in a transform-based audio codec. In such an embodiment,
this existing LPC filter can be directly used for smoothing the discontinuity without the
additional complexity needed to estimate a new LPC filter.
In the following, embodiments of the present invention will be described with reference to
the accompanying drawings, in which:
Fig. 1 shows a block diagram for processing consecutive frames of an audio signal
in accordance with a conventional approach,
Fig. 2 shows a block diagram of another conventional approach for processing
consecutive audio frames for removing a discontinuity,
Fig. 3 shows a simplified block diagram of a system for transmitting audio signals
implementing the inventive approach for removing a discontinuity between
consecutive frames of an audio signal at the encoder side and/or at the
decoder side,
Fig. 4 shows a flow diagram depicting the inventive approach for removing a
discontinuity between consecutive frames of an audio signal in accordance
with an embodiment,
Fig. 5 shows a schematic block diagram for processing a current audio frame in
accordance with embodiments of the present invention avoiding undesired
distortion in the output signal despite the removal of the discontinuities,
Fig. 6 shows a flow diagram representing the functionality of the block in Fig. 5 for
generating the ZIR,
Fig. 7 shows a flow diagram representing the functionality of the block in Fig. 5 for
processing the filtered current frame beginning portion in case the filter block
comprises a recursive filter, like an MR filter, and
Fig. 8 shows a flow diagram representing the functionality of the block in Fig. 5 for
processing the filtered current frame beginning portion in case the filter block
comprises a non-recursive filter, like a FIR filter.
In the following, embodiments of the inventive approach will be described in further detail
and it is noted that in the accompanying drawing elements having the same or similar
functionality are denoted by the same reference signs.
Fig. 3 shows a simplified block diagram of a system for transmitting audio signals
implementing the inventive approach at the encoder side and/or at the decoder side. The
system of Fig. 3 comprises an encoder 200 receiving at an input 202 an audio signal 204.
The encoder includes an encoding processor 206 receiving the audio signal 204 and
generating an encoded audio signal that is provided at an output 208 of the encoder. The
encoding processor may be programmed or built to implement the inventive approach for
processing consecutive audio frames of the audio signal received to avoid discontinuities.
In other embodiments the encoder does not need to be part of a transmission system,
however, it can be a standalone device generating encoded audio signals or it may be
part of an audio signal transmitter. In accordance with an embodiment, the encoder 200
may comprise an antenna 2 10 to allow for a wireless transmission of the audio signal, as
is indicated at 2 12. In other embodiments, the encoder 200 may output the encoded audio
signal provided at the output 208 using a wired connection line, as it is for example
indicated at reference sign 214.
The system of Fig. 3 further comprises a decoder 250 having an input 252 receiving an
encoded audio signal to be processed by the encoder 250, e.g. via the wired line 214 or
via an antenna 254. The encoder 250 comprises a decoding processor 256 operating on
the encoded signal and providing a decoded audio signal 258 at an output 260. The
decoding processor 256 may be implemented to operate in accordance with the inventive
approach on consecutive frames that are filtered in such a way that discontinuities are
avoided. In other embodiments the decoder does not need to be part of a transmission
system, rather, it may be a standalone device for decoding encoded audio signals or it
may be part of an audio signal receiver.
In the following, embodiments of the inventive approach that may be implemented in at
least one of the encoding processor 206 and the decoding processor 256 will be
described in further detail. Fig. 4 shows a flow diagram for processing a current frame of
the audio signal in accordance with an embodiment of the inventive approach. The
processing of the current frame will be described, and the past frame is assumed to be
already processed with the same technique described below. In accordance with the
present invention, in step S 100 a current frame of the audio signal is received. The current
frame is filtered in step S102, for example in a way as described above with regard to
Figs. 1 and 2 (see filter block 102). In accordance with the inventive approach, a
discontinuity between the filtered past frame n-1 and the filtered current frame n (see Fig.
1 or 2) will be removed using linear predictive filtering as is indicated at step S 104. In
accordance an embodiment the linear predictive filter may be defined as
with M the filter order and am the filter coefficients (with a = 1). This kind of filter is also
known as Linear Predictive Coding (LPC). In accordance with embodiments the filtered
current frame is processed by applying linear predictive filtering to at least a part of the
filtered current frame. The discontinuity may be removed by modifying a beginning portion
of the filtered current frame by a signal obtained by linear predictive filtering a predefined
signal with initial states of the linear predictive coding filter defined on the basis of a last
part of the past frame. The initial states of the linear predictive coding filter may be defined
on the basis of a last part of the past frame filtered using the set of filter parameters for the
current frame. The inventive approach is advantageous as it does not require filtering the
current frame of an audio signal with a filter coefficient that is used for the past frame and
thereby avoids problems that arise due to the mismatch of the filter parameters for the
current frame and for the past frame as they are experienced in the prior art approaches
described above with reference to Fig. 2.
Fig. 5 shows a schematic block diagram for processing a current audio frame of the audio
signal in accordance with embodiments of the present invention avoiding undesired
distortion in the output signal despite the removal of the discontinuities. In Fig. 5, the same
reference signs as in Figs. 1 and 2 are used. A current frame n of the audio signal 100 is
received, each frame of the audio signal 100 having a plurality of samples. The current
frame n of the audio signal 100 is processed by the filter block 102. When compared to
the prior art approaches of Figs. 1 and 2, in accordance with embodiments as described
with regard to Fig. 5, the filtered current frame is further processed on the basis of ZIR
samples as is schematically shown by block 110. In accordance with an embodiment on
the basis of the past frame n-1 , and on the basis of an LPC filter the ZIR samples are
produced as is schematically shown by block 112.
The functionality of the processing blocks 110 and 112 will now be described in further
detail. Fig. 6 shows a flow diagram representing the functionality of the processing block
112 for generating the ZIR samples. As mentioned above, the frames of an audio signal
100 are filtered with a linear filter H(z) using filter parameters c selected or determined for
the respective frame. The filter H(z) may be a recursive filer, e.g., an MR filter, or it may be
a non-recursive filter, e.g., a FIR filter. In the processing block 112 a LPC filter is used
which may or may not be quantized. The LPC filter is of the order M and may be either
estimated on the filtered or non-filtered audio signal or may be the LPC filter that is also
used in an audio codec. In a first step S200, the M (M = the order of the LPC filter) last
samples of the past frame n-1 are filtered with the filter H(z) using, however, the filter
parameters or coefficients C of the current frame n. Step S200 thereby produces a first
portion of filtered signal. In step S202 the M last samples of the filtered past frame n-1 (the
M last samples of the past frame filtered using the filter parameters or coefficients c0 of
the past frame n-1) are subtracted from the first portion of filtered signal provided by step
S200, thereby producing a second portion of filtered signal. In step S204 the LPC filter
having the order M is applied, more specifically a zero input response (ZIR) of the LPC
filter is generated in step S204 by filtering a frame of zero samples, wherein the initial
states of the filter are equal to the second portion of filtered signals, thereby generating
the ZIR. In accordance with embodiments, the ZIR can be windowed such that its
amplitude decreases faster to 0.
The ZIR, as described above with regard to Fig. 5, is applied in the processing block 110,
the functionality of which is described with reference to the flow diagram of Fig. 7 for the
case of using, as the linear filer H(z), a recursive filter, like an MR filter. In accordance with
the embodiment described with regard to Fig. 5, to remove discontinuities between the
current frame and the past frame while avoiding undesired distortions, filtering the current
frame n comprises processing (filtering) the current frame n on a sample-by-sample basis,
wherein the samples of the beginning portion are treated in accordance with the inventive
approach. To be more specific, M samples of a beginning portion of the current frame n
are processed, and at a first step S300 the variables m is set to 0. In a next step S302, the
sample m of the current frame n is filtered using the filter H(z) and the filter coefficients or
parameters C for the current frame n. Thus, other than in conventional approaches, the
current frame, in accordance with the inventive approach, is not filtered using coefficients
from the past frame, but only coefficients from the current frame, which as a consequence
avoids the undesired distortion which exist in conventional approaches despite the fact
that discontinuities are removed. Step S302 yields a filtered sample m, and in step S304
the ZIR sample corresponding to sample m is subtracted from the filtered sample m
yielding the corresponding sample of the filtered current frame n. In step S306 it is
determined whether the last sample M of the beginning portion of the current frame n is
processed. In case not all M samples of the beginning portions have been processed, the
variable m is incremented and the method steps S302 to S306 are repeated for the next
sample of the current frame n. Once all M samples of the beginning portions have been
processed, at step S308 the remaining samples of the current frame n are filtered using
the filter parameters of the current frame Ci , thereby providing the filtered current frame n
processed in accordance with the inventive approach avoiding undesired distortion upon
removal of the discontinuities between consecutive frames.
In accordance with another embodiment, the linear filer H(z) is a non-recursive filter, like a
FIR filter, and the ZIR, as described above with regard to Fig. 5, is applied in the
processing block 110. The functionality of this embodiment is described with reference to
the flow diagram of Fig. 8. The current frame n, at step S400, is filtered with the filter H(z)
using the filter coefficients or parameters c for the current frame. Thus, other than in
conventional approaches, the current frame, in accordance with the inventive approach, is
not filtered using coefficients from the past frame, but only coefficients from the current
frame, which as a consequence avoids the undesired distortion which exist in
conventional approaches despite the fact that discontinuities are removed. In step S402 a
beginning portion of the ZIR is subtracted from a corresponding beginning portion of the
filtered current frame, thereby providing the filtered current frame n having the beginning
portion filtered/processed in accordance with the inventive approach and the remaining
part only filtered using filter coefficients or parameters C for the current frame, thereby
avoiding undesired distortion upon removal of the discontinuities between consecutive
frames.
The inventive approach may be applied in situations as described above when the audio
signal is filtered. In accordance with embodiments, the inventive approach may also be
applied at the decoder side, for example, when using an audio codec postfilter for
reducing the level of coding noise between signal harmonics. For processing the audio
frames at the decoder the postfilter, in accordance with an embodiment, may be as
follows:
H(z) = ( 1 - B(z)) / ( 1 - A(z) · z t )
where B(z) and A(z) are two FIR filters and the H(z) filter parameters are the coefficients
of the FIR filters B(z) and A(z), and T indicates the pitch lag. In such a scenario, the filter
may also introduce a discontinuity between the two filtered frames, for example when the
past filter frame parameters c0 are different from the current frame filter parameters Ci ,
and such a discontinuity may produce an artifact in the filtered audio signal 104, for
example a "click". This discontinuity is removed by processing the filtered current frame as
described above in detail.
Although some aspects of the described concept have been described in the context of an
apparatus, it is clear that these aspects also represent a description of the corresponding
method, where a block or device corresponds to a method step or a feature of a method
step. Analogously, aspects described in the context of a method step also represent a
description of a corresponding block or item or feature of a corresponding apparatus.
Depending on certain implementation requirements, embodiments of the invention can be
implemented in hardware or in software. The implementation can be performed using a
digital storage medium, for example a floppy disk, a DVD, a Blue-Ray, a CD, a ROM, a
PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable
control signals stored thereon, which cooperate (or are capable of cooperating) with a
programmable computer system such that the respective method is performed. Therefore,
the digital storage medium may be computer readable.
Some embodiments according to the invention comprise a data carrier having
electronically readable control signals, which are capable of cooperating with a
programmable computer system, such that one of the methods described herein is
performed.
Generally, embodiments of the present invention can be implemented as a computer
program product with a program code, the program code being operative for performing
one of the methods when the computer program product runs on a computer. The
program code may for example be stored on a machine readable carrier.
Other embodiments comprise the computer program for performing one of the methods
described herein, stored on a machine readable carrier.
In other words, an embodiment of the inventive method is, therefore, a computer program
having a program code for performing one of the methods described herein, when the
computer program runs on a computer.
A further embodiment of the inventive methods is, therefore, a data carrier (or a digital
storage medium, or a computer-readable medium) comprising, recorded thereon, the
computer program for performing one of the methods described herein.
A further embodiment of the inventive method is, therefore, a data stream or a sequence
of signals representing the computer program for performing one of the methods
described herein. The data stream or the sequence of signals may for example be
configured to be transferred via a data communication connection, for example via the
Internet.
A further embodiment comprises a processing means, for example a computer, or a
programmable logic device, configured to or adapted to perform one of the methods
described herein.
A further embodiment comprises a computer having installed thereon the computer
program for performing one of the methods described herein.
In some embodiments, a programmable logic device (for example a field programmable
gate array) may be used to perform some or all of the functionalities of the methods
described herein. In some embodiments, a field programmable gate array may cooperate
with a microprocessor in order to perform one of the methods described herein. Generally,
the methods are preferably performed by any hardware apparatus.
The above described embodiments are merely illustrative for the principles of the present
invention. It is understood that modifications and variations of the arrangements and the
details described herein will be apparent to others skilled in the art. It is the intent,
therefore, to be limited only by the scope of the impending patent claims and not by the
specific details presented by way of description and explanation of the embodiments
herein.

CLAIMS
1. A method for processing an audio signal ( 1 00), the method comprising:
removing (S102, S 104, S300-S308, S400-S402) a discontinuity (106a, 106b) between a
filtered past frame and a filtered current frame of the audio signal using linear predictive
filtering.
2. The method of claim 1, comprising filtering the current frame of the audio signal
and removing the discontinuity by modifying a beginning portion of the filtered current
frame by a signal obtained by linear predictive filtering a predefined signal with initial
states of the linear predictive filter defined on the basis of a last part of the past frame.
3. The method of claim 2, wherein the initial states of the linear predictive filter are
defined on the basis of a last part of the unfiltered past frame filtered using the set of filter
parameters for filtering the current frame.
4. The method of one of claims 1 to 3, further comprising estimating the linear
predictive filter on the filtered or non-filtered audio signal ( 1 00).
5. The method of claim 4, wherein estimating the linear predictive filter comprises
estimating the filter based on the past and/or current frame of the audio signal ( 100) or
based on the past filtered frame of the audio signal ( 100) using the Levinson-Durbin
algorithm.
6. The method of one of claims 1 to 3, wherein the linear predictive filter comprises a
linear predictive filter of an audio codec.
7. The method of one of claims 1 to 6, wherein removing the discontinuity comprises
processing the beginning portion of the filtered current frame, wherein the beginning
portion of the current frame has a predefined number of samples being less or equal than
the total number of samples in the current frame, and wherein processing the beginning
portion of the current frame comprises subtracting (S304, S402) a beginning portion of a
zero-input-response (ZIR) from the beginning portion of the filtered current frame.
8. The method of claim 7, comprising filtering (S400) the current frame of the audio
WO 2016/015950 PCT/EP2015/065219
signal using a non-recursive filter, like a FIR filter, for producing the filtered current frame.
9. The method of claim 7, comprising processing the unfiltered current frame of the
audio signal on a sample-by-sample basis using a recursive filter, like an MR filter, and
wherein processing a sample of the beginning portion of the current frame comprises:
filtering (S302) the sample with the recursive filter using the filter parameters of the current
frame for producing a filtered sample, and
subtracting (S304) a corresponding ZIR sample from the filtered sample for producing the
corresponding sample of the filtered current frame.
10. The method of claim 9, wherein filtering (S302) and subtracting (S304) are
repeated until the last sample in the beginning portion of the current frame is processed,
and wherein the method further comprises filtering (S306) the remaining samples in the
current frame with the recursive filter using the filter parameters of the current frame.
11. The method of one of claims 7 to 10, comprising generating the ZIR, wherein
generating the ZIR comprises:
filtering (S200) the M last samples of the unfiltered past frame with the filter and the filter
parameters used for filtering the current frame for producing a first portion of filtered
signal, wherein M is the order of the linear predictive filter,
subtracting (S202) from the first portion of filtered signal the M last samples of the filtered
past frame, filtered using the filter parameters of the past frame, for generating a second
portion of filtered signal, and
generating (S204) a ZIR of a linear predictive filter by filtering a frame of zero samples
with the linear predictive filter and initial states equal to the second portion of filtered
signal.
12. The method of claim 11, comprising windowing the ZIR such that its amplitude
decreases faster to zero.
13. A non-transitory computer program product comprising a computer readable
WO 2016/015950 PCT/EP2015/065219
medium storing instructions which, when executed on a computer, carry out the method of
one of claims 1 to 12.
14. An apparatus for processing an audio signal (100), the apparatus comprising:
a processor ( 102, 110, 112) for removing a discontinuity between a filtered past frame and
a filtered current frame of the audio signal using linear predictive filtering.
15. An apparatus for processing an audio signal (100), the apparatus being configured
to operate according to the method of one of claims 1 to 12.
16. An audio decoder (250), comprising an apparatus of claim 14 or 15.
17. An audio encoder (200), comprising an apparatus of claim 14 or 15.