Resampling output signals of QMF based audio codecs
Specification
The present invention relates to audio processing and, in particular to an apparatus and
method for resampling ouput signals of QMF based audio codecs.
Most low end audio consumer electronics use digital to analogue converters with fixed
sampling rates because of cost reasons. But when multimedia enabled devices are required
to support different kinds of audio sources, the process of resampling is unavoidable,
because media files might be encoded using different sampling rates, and also
communication codecs use different sampling rates. Choosing different sample rates is an
important matter in regard to the operating points of different audio codecs and processing
methods. The more different sample rates that are required to be supported, the more
complex is the sample rate adaption and resampling task.
For example in the current MPEG-D USAC (USAC = Unified Speech and Audio Coding)
reference model, some uncommon (not an integer multiple of 16000Hz or 22050Hz)
sampling rates are employed. These rates are the result of a compromise between two
aspects: First, a nominal sampling rate of the. integrated ACELP coding tool to which it
was specifically designed and which, to a degree, dictates the overall system sampling rate,
and second, the desire to increase the sampling rate together with bit rate to be able to code
greater audio bandwidth and/or to realize scalability.
Partly, the uncommon sampling rates are also a legacy from the AMR-WB+ system which
parts of the reference model have been deduced from. Also, as common in practice in low
bit rate audio coding, the sampling rate and thus the audio bandwidth are being greatly
reduced at low bit rate USAC operating points.
At low USAC bit rates in particular the currently employed sampling rates exhibit both of
the above mentioned problems. They axe not compatible with low-cost hardware D/A
converters and would require an additional post-resampling step. Audio bandwidth is
limited to the Nyquist frequency, which is well below the upper limit of the human audible
range.
To adapt the output sampling rate of an audio processing unit, additional resampling
functional modules are being used for this purpose, requiring a significant amount of
additional computational resources. The technology used for this purpose has not changed
in a lot of time, consisting basically of an interpolator and optional up sampler and down
sampler modules.
It is an object of the present invention to provide an improved concept for resampling
audio signals. The object of the present invention is solved by an apparatus according to
claim I, a method according to claim 14 and a computer program according to claim 15,
According to the present invention, an apparatus for processing an audio signal is
provided. The apparatus comprises a configurable first audio signal processor for
processing the audio signal in accordance with different configuration settings to obtain a
processed audio signal, wherein the apparatus is adapted so that different configuration
settings result in different sampling rates of the processed audio signal. The apparatus
furthermore comprises an analysis filter bank having a first number of analysis filter bank
channels, a synthesis filter bank having a second number of synthesis filter bank channels
and a second audio processor being adapted to receive and process an audio signal having
a predetermined sampling rate.
Moreover, the apparatus comprises a controller for controlling the first number of analysis
filter bank channels or the second number of synthesis filter bank channels m accordance
with a configuration setting provided to the configurable first audio signal processor, so
that an audio signal output of the synthesis filter bank has the predetermined sampling rate
or a sampling rate being different from the predetermined sampling rate and being closer to
the predetermined sampling rate than a sampling rate of an analysis filter bank input signal.
The present invention is based on the finding that by varying the frequency domain
representation signal bandwidth, the equivalent resulting time domain signal will have a
different sampling rate as in the case if no bandwidth change would have been done in
frequency domain. The operation of bandwidth change is cheap, since it can be
accomplished by deleting or adding frequency domain data.
The conversion step from frequency domain back to time domain must be modified in
order to be able to handle the different frequency domain bandwidth (transform length).
The modified bandwidth frequency domain signal representation can also be extended to
the whole signal processing method instead of being limited to the filter bank, thus
allowing the whole process take advantage of the actual target output signal characteristics.
Even if not all audio signal sources can be brought to one single output sample rate,
reducing the amount of different output sample rates already saves a lot of computational
resources on a given device.
The complexity of a filter bank is directly related to its length. If a filter bank time domain
signal synthesis transform is modified for down sampling by reducing the transform
length, its complexity will decrease. If it is used for up sampling by enlarging its transform
length its complexity will increase, but still far below the complexity required for an
additional resampler with equivalent signal distortion characteristics. Also the signal
distortion in total will be less, since any additional signal distortion caused by an additional
resampler will be eliminated.
According to an embodiment, the analysis filter bank is adapted to transform the analysis
filter bank input signal being represented in a time domain into a first time-frequency
domain audio signal having a plurality of fi rst subband signals, wherein the number of first
subband signals is equal to the first number of analysis filter bank channels. According to
this embodiment, the apparatus further comprises a signal adjuster being adapted to
generate a second time-frequency domain audio signal having a plurality of second
subband signals from the first time-frequency-domain audio signal based on the
configuration setting (conf), such that the number of second subband signals of the second
time-frequency domain audio signal is equal to the number of synthesis filter bank
channels. The number of second subband signals of the second time-frequency domain
audio signal is different from the number of subband signals of the first time-frequency
domain audio signal. Furthermore, the synthesis filter bank is adapted to transform the
second time-frequency domain audio signal into a time domain audio signal as the audio
signal output of the synthesis filter bank.
in another embodiment, the signal adjuster may be adapted to generate the second timefrequency
domain audio signal by generating at least one additional subband signal. In a
fuilher embodiment, the signal adjuster is adapted to generate at least one additional
subband signal by conducting spectral band replication to generate at least one additional
subband signal. In another embodiment, the signal adjuster is adapted to generate a zero
signal as an additional subband signal.
According to an embodiment, the analysis filter bank is a QMF (Quadrature Mirror Filter)
analysis filter bank and the synthesis filter bank is a QMF synthesis filter bank in an
alternative embodiment, the analysis filter bank is an MDCT (Modified Discrete Cosine
Transform) analysis filter bank and the synthesis filter bank is an MDCT synthesis filter
bank.
in an embodiment, the apparatus may comprise an additional resampler being adapted to
receive a synthesis filter bank output signal having a first synthesis sampling rate. The
additional resampler may resample the synthesis filter bank output signal to receive a
resampled output signal having a second synthesis sampling rate. By combing the
apparatus according to an embodiment and an additional resampler it is possible to
decrease the complexity of the employed resampler. Instead of employing a highcomplexity
resampler, two low-complexity resampler may be employed.
in another embodiment, the apparatus may be adapted to feed a synthesis filter bank output
signal having a first synthesis sampling rate into the analysis filter bank as an analysis filter
hank input signal. By this, again, the complexity of the apparatus according to an
embodiment may be reduced, instead of employing an analysis filter bank and a synthesis
filter bank having a huge number of analysis and synthesis interbank channels, the number
of filter bank channels will be significantly reduced. This is achieved by repeating the
analysis and synthesis transformations one or more times. According to an embodiment,
the analysis and synthesis filter banks may be adapted such that the number of analysis
and synthesis filter bank channels may be changeable for each transformation cycle (one
transformation cycle comprises an analysis step and a synthesis step).
The controller may be adapted to receive a configuration setting comprising an index
number. Furthermore, the controller may then be adapted to determine the sampling rate of
the processed audio signal or the predetermined sampling rate based on the index number
and a lookup table. According to these embodiments, it is not necessary to transmit the
explicit numbers of analysis and synthesis filter bank channels in each configuration
setting, but instead, a single index number identifying the particular configuration is
transmitted. This reduces the bit rate needed for transmitting a configuration setting.
According to an embodiment, the controller is adapted to determine the first number of
analysis filter bank channels or the second number of synthesis filter bank channels based
on a tolerable error. In an embodiment, the controller may comprise an error comparator
for comparing the actual error with a tolerable error. Furthermore, the apparatus may be
adapted to obtain the tolerable error from the configuration setting. According to these
embodiments, it may be possible to specify the degree of accuracy of the resampling, it
may be appreciated that in certain situations, the accuracy of the resampling can be
reduced to also reduce on the other hand the complexity of the analysis and synthesis filter
bank and thus to reduce the complexity of the calculation.
According to another embodiment, an apparatus for upmixing a surround signal is
provided. The apparatus comprises an analysis filter bank for transforming a downmixed
time domain signal into a time-frequency domain to generate a plurality of downmixed
subband signals. Moreover, the apparatus comprises at least two upmix units for upmixing
the plurality of subband signals to obtain a plurality of surround subband signals.
Furthermore, the apparatus comprises at least two signal adjuster units for adjusting the
number of surround subband signals. The at least two signal adjuster units are adapted to
receive a first plurality of input surround subband signals. The at least two signal adjuster
units are adapted to output a second plurality of output surround subband signals, and
wherein the number of the first plurality of input surround subband signals and the number
of the second plurality of output surround subband signals is different. Moreover, the
apparatus comprises a plurality of synthesis filter bank units for transforming a plurality of
output surround subband signals from a time-frequency domain to a time domain to obtain
time domain surround output signals. Furthermore, the apparatus comprises a controller
which is adapted to receive a configuration setting. The controller is moreover adapted to
control the number of channels of the analysis filter banks to control the number of
channels of the synthesis filter bank units, to control the number of the first plurality of
input surround subband signals of the signal adjuster units, and to control the number of
the second plurality of output surround subband signals of the signal adjuster units based
on the received configuration setting.
Preferred embodiments of the present invention are subsequently discussed with respect to
the accompanying figures, in which:
Fig. 1 illustrates an apparatus for processing an audio signal according to
an embodiment,
Figs, 2a - 2c depict the transformation of time domain samples into timefrequency
domain samples.
Fig. 3a 3b illustrate the transformation of time-frequency domain samples into
time domain samples,
Fig. 4 depict in a further illustration the transformation of time-frequency
domain samples into time domain samples,
illustrate two diagrams depicting a basic concept of an embodiment,
illustrates an apparatus according to a further embodiment,
show lookup tables in accordance with an embodiment,
illustrates an apparatus according to an embodiment employing SBR
processing,
depicts an apparatus according to another embodiment employing
QMF analysis and synthesis filter banks for upmixing an MPEG
Surround signal with a resampled sampling rate according to an
embodiment
illustrates an apparatus according to another embodiment employing
SBR processing,
depicts an apparatus according to another embodiment comprising
an additional resampler,
illustrates an apparatus employing QMF as resampler according to
an embodiment,
shows an apparatus employing an additional resampler according to
an embodiment,
illustrates an apparatus employing QMF as resampler according to
another embodiment,
depicts an apparatus according to a further embodiment wherem the
apparatus is adapted to feed the synthesis filter bank output into the
analysis filter bank to conduct another transformation cycle,
Fig, 16 illustrates a controller according to another embodiment comprising
an error comparator,
Fig. 17 shows a flow chart depicting a method for determining the number
of analysis and synthesis filter bank channels, respectively, and
Fig. illustrates a controller according to a further embodiment comprising
an error comparator.
Fig. 1 illustrates an apparatus for processing an audio signal according to an embodiment.
An audio signal so is fed into the apparatus, in another embodiment. so may be a bit stream,
in particular an audio data bit stream, Moreover, the apparatus receives a configuration
setting conf. The apparatus comprises a configurable first audio signal processor 110 for
processing the audio signal so in accordance with the configuration setting conf to obtain a
processed audio signal s\ . Furthermore, the apparatus for processing an audio signal is
adapted so that different configuration settings conf result in different sampling rates of the
processed audio signal. The apparatus furthermore comprises an analysis filter bank 120
having a first number of analysis filter bank channels c5 and a synthesis filter bank 130
having a second number of synthesis filter bank channels <¾, Moreover, the apparatus
comprises a second audio processor 140 being adapted to receive and process an audio
signal s2 having a predetermined sampling rate. Furthermore, the apparatus comprises a
controller 150 for controlling the first number of analysis filter bank channels Cj or the
second number of synthesis filter bank channels c2 in accordance with a configuration
setting conf provided to the configurable first audio signal processor 110, so that an audio
signal output s2 by the synthesis filter bank 130 has the predetermined sampling rate or a
sampling rate being different from the predetermined sampling rate, but which is closer to
the predetermined sampling rate than the sampling rate of an input signal s ¾ into the
analysis filter bank 120.
The analysis filter bank and the synthesis filter bank might be adapted such that the
number of analysis channels and the number of synthesis channels are configurable and
that their number might be determined by configurable parameters.
in Figs, 2a - 2c, the transformation of time domain samples into time-frequency domain
samples is illustrated. The left side of Fig. 2a illustrates a plurality of samples of a
(processed) audio signal in a time domain. On the left side of Fig. 2a, 640 time samples are
illustrated (the latest 64 time samples are referred to as "new time samples" while the
remaining 576 time samples are referred to as old time samples, in the embodiment
depicted by Fig. 2a, a first step of a Short Time Fourier Transform (STFT) is conducted.
The 576 old time samples and the 64 new time samples are transformed to 64 frequency
values, i.e. 64 subband sample values are generated.
In a subsequent step illustrated in Fig. 2b, the oldest 64 time samples of the considered 640
time samples are discarded. Instead, 64 new time samples are considered together with the
remaining 576 already considered time samples available in the processing step illustrated
by Fig, 2a. This could be regarded as shifting a sliding window having a length of 640 time
samples by 64 time samples in each processing step. Again, also in the processing step
depicted in Fig. 2b, further 64 subband samples are generated from the considered 640
time samples (576 old time samples and 64 new time samples considered for the first
time). By this, a second set of 64 subband values is generated. One could say that 64 new
subband samples are generated by taking 64 new time samples into account.
In the subsequent step depicted in Fig. 2c, again, the sliding window is shifted by 64 time
samples, i.e. the oldest 64 time values are discarded and 64 new time samples are taken
into account. 64 new subband samples are generated based on the 576 old time samples
and 64 new time samples. As can been seen in Fig. 2c, right side, a new set of 64 new
subband values has been generated by conducting STFT.
The process illustrated in Figs. 2a - 2c is conducted repeatedly to generate additional
subband samples from additional time samples.
Explained in general terms, 64 new time samples are needed to generate 64 new subband
samples.
In the embodiment illustrated by Figs. 2a - 2c, each set of the generated subband samples
represents the subband samples at a particular time index in a time-frequency domain, i.e.,
the 32nd subband sample of time index j represents a signal sample S[32,j] in a timefrequency
domain. Regarding a certain time index in the time-frequency domain, 64
subband values exist for that time index, while for each point-in-time in the time domain,
at most a single signal value exist. On the other hand, the sampling rate of each of the 64
frequency bands is only 1/64 of the signal in the time-domain.
It is understood by a person skilled in the art that the number of subband signals, which are
generated by an analysis filter bank depends on the number of channels of the analysis
filter bank. For example, the analysis filter bank might comprise 16, 32, 96 or 128
channels, such that 16, 32, 96, or 128 subband signals in a time frequency domain might be
generated from e.g. 16, 32, 96 or 128 time samples, respectively.
Fig. 3a - 3b illustrate the transformation of time-frequency domain samples into time
domain samples:
The left side of Fig, 3a illustrates a plurality of sets of subband samples in a timefrequency
domain in more detail, each longitudinal box in Fig. 3a represents a plurality of
64 subband samples in a time-frequency domain. A sliding window in the time-frequency
domain covers 10 time indexes each comprising 64 subband samples in the time-frequency
domain. By conducting an inverse Short Time Fourier Transform (ISTFT), 64 time
samples are generated from the considered (10 times 64) subband samples, as depicted in
Fig. 3a, right side.
In a subsequent processing step illustrated in Fig. 3b, the oldest set of 64 subband values is
discarded, instead, the sliding window now covers a new set of 64 subband values having a
different time index in the time-frequency domain, 64 new time samples are generated in
the time domain from the considered 640 subband samples (576 old subband samples and
64 new subband samples considered for the first time). Fig. 3b, right side illustrates the
situation in the time domain. Fig. 3b depicts 64 old time samples generated by conducting
the ISTFT as illustrated in Fig, 3a are depicted together with the 64 new time samples
generated in the processing step of Fig, 3b,
The process illustrated in Figs, 3a - 3b is conducted repeatedly to generate additional time
samples from additional subband samples.
To explain the concept of the synthesis filter bank 130 in general terms, 64 new subband
samples in a time-frequency domain are needed to generate 64 new time samples in a time
domain.
It is understood by a person skilled in the art, that the number of time samples which are
generated by a synthesis filter bank depends on the number of channels of the synthesis
filter bank. For example, the synthesis filter bank might comprise 16, 32, 96 or 128
channels, such that 16, 32, 96, or 128 time samples in a time domain might be generated
from e.g. 16, 32, 96 or 128 subband samples in a time-frequency domain, respectively.
Fig, 4 presents another illustration depicting the transformation of time-frequency domain
samples into time domain samples, In each processing step, an additional 64 subband
samples are considered (i.e. the 64 subband samples of the next time index in a timefrequency
domain). Taking the latest 64 subband samples into account, 64 new time
samples can be generated The sampling rate of the signal in the time domain is 64 times
the sampling rate of each one of the 64 subband signals.
Fig. 5 illustrates two diagrams depicting a basic concept of an embodiment, The upper part
of Fig. 5 depicts a plurality of subband samples of a signal in a time-frequency domain.
The abscissa represents time. The ordinate represents frequency. Fig. 5 differs from Fig. 4
in that for each time index, the signal in the time-frequency domain contains three
additional subband samples (marked with "x"). i.e. the three additional subbands have
been added such that the signal in the time-frequency domain does not only have 64
subband signals, but now does have 67 subband signals. The diagram illustrated at the
bottom of Fig, 5 illustrates time samples of the same signal in the time domain after
conducting an inverse Short Time Fourier Transform (iSTFT), As 3 subbands have been
added in the time-frequency domain, the 67 additional subband samples of a particular
time index in the time-frequency domain can be used to generate 67 new time samples of
the audio signal in the time domain. As new 67 time samples have been generated in the
time domain using the 67 additional subband samples of a single time index in the timefrequency
domain, the sampling rate of the audio signal s2 in the time domain as outputted
by the synthesis filter bank 130 is 67 times the sampling rate of each one of the subband
signals. As could be seen above, employing 64 channels in the analysis filter bank 120
results in a sampling rate of each subband signal of 1/64 of the sampling rate of the
processed audio signal si as fed into the analysis filter bank 120, Regarding the analysis
filter bank 120 and the synthesis filter bank Ϊ 30 together, the analysis filter bank 120
having 64 channels and the synthesis filter bank 130 having 67 channels results in a
sampling rate of the signal s2 outputted by the synthesis filter bank of 67/64 times the
sampling rate of the audio signal si being inputted into the analysis filter bank 120.
The following concept can be derived: Consider a (processed) audio signal si that is fed
into the analysis filter bank 120. Assuming that the filter bank has Ci channels and,
assuming further that the sampling rate of the processed audio signal is sr , then the
sampling rate of each subband signal is Assuming further that the synthesis filter
bank has c2 channels and assuming that the sampling rate of each subband signal is
srSuhhand , then the sampling rate of the audio signal s2 being outputted by the synthesis filter
bank 130 is c2 . srSuhhand - That means, the sampling rate of the audio signal being outputted
by the synthesis filter bank 130 is c2/ c1 . sr1. Selecting c2 different from Ci means that the
sampling rate of the audio signal s2 being outputted by the synthesis filter bank 130 can be
set differently from the sampling rate of the audio signal being inputted into the analysis
filter bank 120.
Choosing c2 different from c\ does not only mean that the number of analysis filter bank
channels differs from the number of synthesis filter bank channels. Moreover the number
of subband signals being generated by the analysis filter bank 120 by the STFT differs
from the number of subband signals that are needed when conducting the ISTFT by the
synthesis filter bank 130.
Three different situations can be distinguished:
if Ci is equal to c2, the number of subband signals that are generated by the analysis filter
bank 120 is equal to the number of subband signals needed by the synthesis filter bank 130
for the ISTFT. No subband adjustment is needed.
if e2 is smaller than cl s the number of subband signals generated by the analysis filter bank
120 is greater than the number of subband signals needed by the synthesis filter bank 130
for synthesis. According to an embodiment, the highest frequency subband signals might
be deleted. For example, if the analysis filter bank 120 generates 64 subband signals and if
the synthesis filter bank 130 only needs 6 1 subband signals, the three subband signals with
the highest frequency might be discarded,
If c2 is greater than cl s tlien the number of subband signals generated by the analysis filter
bank 120 is smaller than the number of subband signals needed by the synthesis filter bank
130 for synthesis.
According to an embodiment, additional subband signals might be generated by adding
zero signals as additional subband signals. A zero signal is a signal where the amplitude
values of each subband sample are equal to zero.
According to another embodiment, additional subband signals might be generated by
adding pseudorandom subband signals as additional subband signals, A pseudorandom
subband signal is a signal where the values of each subband sample comprise
pseudorandom data, wherein the pseudorandom data has to be detennined pseudorandomly
from an allowed value range. For example, the pseudorandomly chosen amplitude values
of a sample have to be smaller than a maximum amplitude value and the phase values of a
sample have to be in the range between 0 and 2p (inclusive).
in another embodiment, additional subband signals might be generated by copying the
sample values of the highest subband signal and to use them as sample values of the
additional subband signals. In another embodiment, the phase values of the highest
subband are copied and used as sample values for an additional subband, while the
amplitude values of the highest subband signal are multiplied with a weighting factor, e.g.
to decrease their weight and are then used as amplitude values of the subband samples of
the additional subband signal. For example, all amplitude values in an additional subband
signal might be multiplied with the weighting factor 0.9, If two additional subband signals
are needed, the amplitude values of the highest subband signal might be multiplied with a
weighting factor 0.9 to generate a first additional subband signal, while all amplitude
values might be multiplied with a weighting factor 0.8 to generate a second additional
subband signal.
Most highly efficient audio codecs use parametric signal enhancements, which in turn
frequently use a QMF (Quadrature Mirror Filter) (i.e. MPEG-4 HE- AAC), where the
concepts proposed in the above-described embodiments may also be employed, QMF
based codecs use typically a band polyphase filter structure to convert sub
bands into a time domain output signal of a nominal sampling frequency fs, nominal- By
changing the amount of output bands, by adding sub bands containing a zero signal, or
removing some of the higher bands (which might be empty anyway), the output sampling
fs rate can be changed in steps of Af3 as shown below.
which results in an overall output sampling frequency fs of:
Instead of adding an extra sampling rate converter, this functionality can be built into the
already existing QMF' synthesis filter.
The workload increase is below that of a sampling rate converter with comparable
accuracy, but the sampling rate ratio cannot be arbitrary. Essentially it is determined by the
ratio of the number of bands used in the QMF analysis and QMF synthesis filter bank.
Generally it is preferred to use a number of output bands that allows a fast computation of
the synthesis QMF, e.g. 60, 72, 80s 48, ...
The same way as the output sample rate can be changed when employing QMF, the same
way can the sample rate of a audio signal codec be adjusted, which uses another kind of
filter batik, for example a MDCT (Modified Discrete Cosine Transform).
Fig. 6 illustrates an apparatus according to an embodiment The apparatus comprises a
signal adjuster 125, An analysis filter bank 120 is adapted to transform the analysis filter
bank input signal si being represented in a time-domain into a first time-frequency domain
audio signal having a plurality of, e.g., 3 first subband signals sn, s12 , s13 . The number of
first subband signals is equal to the first number c3 of analysis filter bank channels,
The signal adjuster 125 is adapted to generate a second time-frequency domain audio
signal from the first time-frequency domain audio signal based on the configuration setting
conf. The second time-frequency domain audio signal has a plurality of, e.g., 4 second
subband signals s2 1, s22, s23s s2 4 . The second time-frequency domain audio signal is
generated such that the number of second subband signals is equal to the number c2 of
synthesis filter bank channels. The number of second subband signals of the second timefrequency
domain audio signal may be different from the number of subband signals of the
first time-frequency domain audio signal Therefore, the number of subband signals may
have to be adjusted, e.g. according to one of the above-described concepts.
The synthesis filter bank 130 is adapted to transform the second time-frequency domain
audio signal into a time-domain audio signal as the audio signal output s2 of the synthesis
filter bank 130.
However, in other embodiments, a signal adjuster 125 may not be comprised. If the
analysis filter bank 120 provides more channels than needed by the synthesis filter bank
130, the synthesis filter bank may itself discard channels that are not necessary.
Furthermore, the synthesis filter bank 130 may be configured to itself use a zero subband
signal or a signal comprising pseudorandom data, if the number of subband signals
provided by the analysis filter bank 120 is smaller than the number of synthesis filter bank
channels,
The apparatus according to the embodiment is particularly suitable for adapting to different
situations. For example, the first audio signal processor 110 might need to process the
audio signal s0 such that the processed audio signal si has a first sampling rate sri in one
situation and such that the processed audio signal sj has a second sampling rate sr¾ ' being
different from the first sampling rate in a second situation. For example, the first audio
signal processor 110 might employ an ACELP (Algebraic Code Excited Linear Prediction)
decoding tool working with a first sampling rate of e.g. 16000 Hz while in a different
second situation the first audio signal processor might employ an AAC (Advanced Audio
Coding) decoder, e.g. having a sampling rate of e.g. 48000 Hz. Furthermore, the situation
might arise that the first audio signal processor employs an AAC decoder which switches
between different sampling rates. Or, the first signal processor 110 might be adapted to
switch between a first stereo audio signal si having a first sampling rate sr} and a second
audio si' signal being an MPEG Surround signal having a second sampling rate sri'.
Moreover, it might be necessary to provide an audio signal to the second audio signal
processor 140 having a certain predetermined sampling rate s¾. For example, a digital to
analogue converter employed might require a certain sampling rate, in this case, the second
signal processor 140 might always work with a fixed second sampling rate sr2. However, in
other cases, sampling rates of the audio signal s2 at the second audio processor 140 might
change at run time. For example, in a first case, the second audio signal processor 140
might switch between a first low audio quality D/A (digital to analogue) converter
supporting a relatively low sampling rate of e.g. 24000 Hz, while in other situations the
second audio signal processor 140 might employ a second D/A converter having a
sampling rate of e.g. 96000 Hz. For example, in situations where the original sampling rate
of the processed audio signal sr2 having been processed by the first audio signal processor
110 has a relatively low sampling rate of e.g. 4000 Hz it might not be necessary to employ
the high-quality second D/A converter having a sampling rate of 96000 Hz, but instead, it
is sufficient to employ the first D/A converter which requires fewer computational
resources. It is therefore appreciated to provide an apparatus with adjustable sampling
rates.
According to an embodiment, an apparatus is provided which comprises a controller 150
which controls the first number of analysis filter bank channels c and/or the second
number of synthesis filter bank channels c2 in accordance with a configuration setting conf
provided to the configurable first audio signal processor 110, so that an audio signal output
by the synthesis filter bank 130 has the predetermined sampling rate sr2 or a sampling rate
Sr2 being different from the predetermined sampling rate sr2s but being closer to the
predetermined sampling rate sr2 than the sampling rate sri of a processed input signal sj
into the analysis filter bank 120.
In an embodiment, the configui-ation setting might contain an explicit information about
the first sampling rate sri and/or the second sampling rate sr2. For example, the
configuration setting might explicitly define that a first sampling rate sri is set to 9000 Hz
and that a second sampling rate sr2 is set to 24000 Hz.
However, in another embodiment, the configuration setting conf may not explicitly specify
a sampling rate, instead, an index number might be specified which the controller might
use to determine the first Sr 1 and/or the second sampling rate sr2.
In an embodiment, the configuration setting conf may be provided by an additional unit
(not shown) to the controller at run time. For example, the additional unit might specify in
the configuration setting conf, whether an ACELP decoder or an AAC decoder is
employed.
In an alternative embodiment, the configuration setting conf is not provided at run-time by
an additional unit, but the configuration setting conf is stored once such that it is
permanently available for a controller 150, The configuration setting conf then remains
unaltered for a longer time period,
Depending on this determination, the additional unit may send the explicit sampling rates
to the controller being comprised in the configuration setting conf.
In an alternative embodiment, the additional unit sends a configuration setting conf which
indicates whether a first situation exists (by transmitting an index value "0": indicating
"ACELP decoder used", or by transmitting an index value "1": indicating "AAC decoder
used"). This is explained with reference to Fig, 7a and 7b:
Figs, 7a and 7b illustrate lookup tables according to an embodiment being available to a
controller. For example, the lookup table may be predefined lookup table being stored as a
fixed table in the controller. In another embodiment, the lookup table may be provided as
meta information from an additional unit. While, for example, the lookup table information
is only sent once for a long period of time, an index value specifying the current sampling
rate configuration is more frequently updated.
Fig. 7a depicts a simple lookup table allowing the resolution of a single sampling rate, in
the embodiment of Fig. 7a a sampling rate of the first audio signal processor 110 is
specified. By receiving an index value being comprised in the first configuration setting
conf, the controller 150 is able to determine the sampling rate of the processed audio signal
S i being processed by the first audio signal processor 110, in the lookup table of Fig, 7a, no
information about the second sampling rate sr2 is available. In an embodiment, the second
sampling rate is a fixed sampling rate and is known by the controller 150. in another
embodiment, the second sampling rate is determined by employing another lookup table
being similar as the lookup table illustrated in Fig. 7a.
Fig. 7b illustrates another lookup table which comprises information about the first
sampling rate sr1 of the processed audio signal s1 as well as the second sampling rate sr2 of
the audio signal s2 being outputted by the synthesis filter bank. An additional unit transmits
a configuration setting conf comprising an index value to the controller 150. The controller
150 looks up the index value in the lookup table of Fig. 7b and thus determines the first
desired sampling rate of the processed audio signal s i and the second desired sampling rate
sr2 of the audio signal s2 being generated by the synthesis filter bank 140.
Fig, 8 illustrates a combination of the above-described concepts with SBR processing. If
the QMF synthesis band is part of an SBR module, the resampling functionality can be
integrated into the system, in particular, it is then possible to transmit SBR parameters to
extend the active SBR range beyond the usual 2:1 or 4:1 resampling ratio with the
additional merit that it is possible to realize almost arbitrary resampling ratios by
adequately choosing the appropriate M and N of the QMF filter banks, thus increasing the
degrees of freedom for overall resampling characteristic (see Fig. 8).
For example, if the number of synthesis bands is higher than 64, they do not necessarily
have to be filled with zeros. Instead, the range for the SBR patching could also be extended
in order to make use of this higher frequency range ,
in Fig. 8, the resulting QMF output sampling frequency is:
E.g. in case of the USAC 8kbps operation test point, the internal sampling frequency fs,Core
is typically chosen to be 9,6kHz, While sticking to the M=32 band QMF analysis filter
bank, the synthesis could be replaced by an N=80 band QMF bank. This would result in an
output sampling frequency of
By doing so, the potential audio bandwidth which can be covered by SBR can be increased
to 12kHz. At the same time a potential post-resampling step to a convenient 48kHz can be
implemented rather cheaply because the remaining resampling ratio is a simple 1:2
relation.
Many more combinations are conceivable which could allow a wide(r) SBR range while
maintaining the possibility to allow the core coder to run on somewhat unusual or
uncommon sampling frequencies.
Fig. 9 illustrates an apparatus according to another embodiment employing QMF anaiysis
and synthesis filter banks for upmixing an MPEG Surround signal with a resampled
sampling rate according to an embodiment. For illustrative purposes, the analysis filter
bank is depicted to generate only 3 subband signals from the inputted signal and each one
of the QMF synthesis filter banks is depicted to transform a time-frequency domain signal
comprising only four subband signals back to the time domain. However, it is understood
that in other embodiments, the anaiysis interbank might, for example, comprise 45
channels and the synthesis filterbank might, for example, comprise 60 channels,
respectively.
In Fig. 9, a downmixed audio signal S 1 is fed into a QMF analysis filter bank 910. The
QMF analysis filter bank 910 transforms the downmixed time domain audio signal into a
time-frequency domain to obtain three (downmixed) subband signals S11, S12 , S13 . The three
downmixed subband signals S11 , S12, S13 are then fed into three upmix units 921, 922, 923,
respectively. Each one of the upmix units 921, 922, 923 generates five surround subband
signals as a left, right, center, left surround and right surround subband signal, respectively.
The three generated left subband signals are then fed into a left signal adjuster 93 1 for the
left subband signals. The left signal adjuster 931 generates four left subband signals from
the three left surround subband signals and feeds them into a left synthesis filter bank 941
which transforms the subband signals from the time-frequency domain to the time domain
to generate a left channel s2 1 of the surround signal in a time domain, in the same way, a
right signal adjuster 932 and a right synthesis filter bank 942 is employed to generate a
right channel s22, a center signal adjuster 933 and a center synthesis filter bank 943 is
employed to generate a center channel d23, a left surround signal adjuster 934 and a left
surround svntliesis filter bank 944 is employed to generate a left surround channel s24, and
a right surround signal adjuster 935 and a right surround synthesis filter bank 945 is
employed to generate a right surround channel s25 of the surround signal in the time
domain.
A controller (950) receives a configuration setting conf and is adapted to control the
number of channels of the analysis filter bank 910 based on the received configuration
setting conf. The controller is further adapted to control the number of channels of the
synthesis filter bank units 941, 942, 943, 944, 945, the number of the first plurality of input
surround subband signals of the signal adjuster units 931, 932, 933, 934, 935 and the
number of the second plurality of output surround subband signals of the signal adjuster
units 93 1, 932, 933, 934, 935 based on the received configuration setting conf
5 Fig. 10 illustrates an apparatus according to another embodiment. The embodiment of Fig
10 differs from the embodiment of Fig. 8 in that the signal adjuster 125 further comprises a
spectral band replicator 128 for conducting a spectral band replication (SBR) of the
subband signals derived from the analysis filter bank 120 to obtain additional subband
signals.
10
Conventionally, by conducting spectral band replication a plurality of subband signals is
"replicated" such that the number of subband signals derived from the spectral band
replication is twice or four times the number of the subband signals available for being
spectrally replicated. In a conventional spectral band replication (SBR), the number of
15 available subband signals is replicated so that e.g. 32 subband signals (resulting from an
analysis filter bank tansformation) are replicated and such that 64 subband signals are
available for the synthesis step. The subband signals are replicated such that the available
subband signals form the lower subband signals, while the spectrally replicated subband
signals from the higher subband signals being located in frequency ranges higher than the
20 already available subband signals.
According to the embodiment depicted in Fig, 10, the available subband signals are
replicated such that the number of subband signals resulting from SBR does not have to be
an integer multiple of (or the same number as) the replicated subband signals. For example,
25 32 subband signals might be replicated such that not 32 additional subband signals are
derived, but, for example, 36 additional subband signals are derived and that in total, for
example, 68 instead of 64 subband signals are available from synthesis. The synthesis filter
bank 130 of the embodiment of Fig. 10 is adjusted to process 68 channels instead of 64.
30 According to the embodiment illustrated in Fig, 10, the number of channels that are
replicated by the spectral band replication and the number of channels that can be
replicated is adjustable such that the number of replicated channels does not have to be an
integer multiple of (or the same number as) the channels used in the spectral band
replications, in the embodiment of Fig. 10. the controller not only controls the number of
35 channels of the synthesis filter bank 140, but does also control the number of channels to
be replicated by the spectral band replication. For example, if the control ler has determined
that the analysis filter bank 120 has ci channels and the synthesis filter bank has 0 2
channels (c2 > cj), then the number of additional channels that have to be derived by the
spectral band replication is c2 - c1.
if c2 > 2 . C1 , the question arises how to generate additional subband signals in the context
of a spectral band replication. According to an embodiment, a zero subband signal (the
amplitude values of all subband samples are zero) may be added for each additionally
required subband signal. In another embodiment, pseudorandom data is used as sample
values of the additional subband signals to be generated. In a further embodiment, the
highest subband signal resulting from the spectral band replication is itself replicated: For
example, the amplitude values of the highest subband signals are duplicated to form the
amplitude values of the additional one or more subband signal. The amplitude values might
be multiplied by a weighting factor. For example, each one of the amplitude values of the
first additional subband signal might be multiplied by 0.95. Each one of the amplitude
values of the second additional subband signal might be multiplied by 0.90, etc.
In a still further embodiment, the spectral band replication is extended to generate
additional subband signals. Spectral envelope information might be used to generate
additional subband signals from the available lower subband signals. The spectral envelope
information might be used to derive weighting factors used to be multiplied by the
amplitude values of the lower subband signals considered in the spectral band replication
to generate additional subband signal
Fig. 11 illustrates an apparatus according to another embodiment. The apparatus differs
from the apparatus illustrated in Fig. 1 in that the apparatus of Fig. 11 further comprises an
additional resampler 170. The additional resampler 170 is used to conduct an additional
resampling step. The resampler may be a conventional resampler or may alternatively be
an apparatus for processing an audio signal which conducts resampling according to the
invention, if, for example an apparatus according to the invention is used as additional
resampler, the first apparatus according to the invention resampies an audio signal having a
first sampling rate sri to a sampling rate sr2 = c2/ c1 · sr 1. Then, the additional resampler
resampies the audio signal from a sampling rate sr2 to a sampling rate sr2' - c4/c3 · sr2 - c4/
c3 · c2/ c1 · sr1. By employing two resamplers, it is avoided that a resampler according to
one of the above-described embodiments has to have c1 · c3 analysis channels and c4 - c2
synthesis channels. For example, if a resampling factor of 998000/996003 is desired (the
resampling factor is the ratio of the sampling rate of the audio signal after synthesis to the
sampling rate of the audio signal before analysis), then, an apparatus comprising two
resamplers avoids thai 996003 analysis filter bank channels and 998000 synthesis filter
bank channels are needed, instead, a first resampling may be conducted by an analysis
filter bank having 999 fiter bank channels and a synthesis filter bank having 1000 channels
and a second resampling may be conducted by an analysis filter bank having 997 channels
and a synthesis filter bank having 998 channels,
In the embodiment, the controller 150 may be adapted to steer how to split the resampling
factor into suitable analysis and synthesis filter bank channel values.
Fig, 12 illustrates QMF as resampler according to an embodiment. An example of a QMF
synthesis stage with attached post-resampler to adjust the QMF output sampling rate is
depicted.
i f the output sampling rate after QMF synthesis does not comply to a "standard" sampling
rate, a combination of QMF based resampling and an additional resampler can still be used
in order to achieve better operating conditions for a resampler in case this is required (e.g.
benign small integer resampling ratio (or interpolate between near sampling rates, for
example employing a Lagrange interpolator),
in Fig. 13, a resampler is depicted comprising an analysis unit and a synthesis unit, But
since such building blocks are already present in most current audio codecs, these already
existing building blocks can be slightly changed, by means of a controlling entity, in order
to accomplish the resampling task, without requiring additional analysis / synthesis stages
appended to the decoder system. This approach is shown in Fig. 14. in some systems it
might be possible to slightly change fs in order to achieve more convenient operating
points and overcome implementation constraints in regard to the overall decimation and
upsampling factors.
The "Filter bank control" block shown in Figure 13 will manipulate the factors M and N of
the decoder in order to obtain the desired output sampling frequency f s, a.. it takes as
inputs the desired output sampling frequency fs,final, the core decoder output sampling
frequency and other knowledge about the decoder. The sampling frequency f s, a . may b e
desired to be constant, and to match the output device hardware, while from the codec
perspective it might be desirable to change because of coding efficiency aspects. Bymerging
the resampler into the decoder both requirements, a fixed output sampling rate at
the output and best operating sampling rate of the audio codec can be met with almost no
additional complexity and no signal degradation because of additional resampler
processing.
The QMF prototype for the different lengths can be created from the one for the 64 band
QMF by interpolation.
The complexity of a filter bank is directly related to its length. If a filter bank time domain
signal synthesis transform is modified for downsampling by reducing the transform length,
its complexity will decrease. If it is used for upsampling by enlarging its transform length
its complexity will increase, but still far below the complexity required for an additional
resampler with equivalent signal distortion characteristics,
Fig. 15 illustrates an apparatus according to a further embodiment wherein the apparatus is
adapted to feed a synthesis filter bank output into an analysis filter bank to conduct another
transformation cycle. As in the embodiment of Fig 1, a processed audio signal S 1 is fed
into an analysis filter bank 120 where the audio signal is transferred from a time-domain
into a time-frequency domain. The synthesis filter bank then transforms the time-frequency
domain signal back to the time domain, wherein the number of synthesis filter bank
channels c2 is different from the number of analysis filter bank channels c1 to generate an
output signal s2 with a different sampling rate than the inputted signal. Contrary to the
embodiment of Fig. 1, however, the output signal may not be fed into the second audio
signal processor 140, but instead, may be fed again into an analysis filter bank to conduct
an additional resampling of the audio signal by an analysis filter bank and a synthesis filter
bank. Different analysis filter banks and synthesis filter banks (e.g. analysis filter bank
instances and synthesis filter bank instances) may be employed in subsequent
analysis/synthesis steps. The controller 150 may control the number of analysis and
synthesis filter bank channels C1 e2, such that the numbers are different in the second
analysis/synthesis step than in the first analysis/synthesis step. By this the total resampling
ratio may be any be arbitrarily chosen such that it results to (c2 · c4 · eg · c 8 . ...) / (c1
. c3
.
c5. c7 · ,.,), wherein c1, e2, e3 ... are integer values.
Resampling an audio signal having a first sampling rate srj such that it has a second
sampling rate s¾ after resampling might not be easy to realize. For example, in case that a
sampling frequency of 22050 Hz shall be resampled to a sampling frequency of 23983 Hz,
it would be computationally expensive to realize an analysis filter bank having 22050
channels and a synthesis filter bank having 23983 channels. However, although it might be
desirable to exactly realize the output sampling frequency of 23983 Hz the user (or another
application) might tolerate an error as long as the error is within acceptable bounds.
Fig. 16 illustrates a controller according to another embodiment. A first sampling rate sri
and a second desired sampling rate sr2 are fed into the controller, The first sampling rate
specifies the sampling rate of a (processed) audio signal s 1 that is fed into an analysis filter
hank. The second desired sampling rate sr2 specifies a desired sampling rate that the audio
signal s2 shall exhibit when being outputted from a synthesis filter bank. Furthermore, a
tolerable error e is also fed into the controller. The tolerable error e specifies to what
degree an actual sampling rate sr ' of a signal outputted from the synthesis filter bank
might deviate from the desired sampling rate sr2.
The first sampling rate srj and the second desired sampling rate sr2 are fed into a synthesis
channel number chooser 1010. The synthesis channel number chooser 1010 chooses a
suitable number of channels c2 of the synthesis filter bank. Some numbers of synthesis
filter bank channels c2 might be particularly suitable to allow fast computation, of the
signal transformation from a time-frequency domain to a time domain, e.g. 60, 72, 80 or 48
channels. The synthesis channel number chooser 1010 might choose the synthesis channel
number c2 depending on the first and second sampling rate sr1, sr2. For example, if the
resampling ratio is an integer number, for example 3 (resulting e.g. from sampling rates sr
- 16000 Hz and s r 2=48000 Hz), it might be sufficient that the synthesis channel number is
a small number, e.g. 30. in other situations it might be more useful to choose a bigger
synthesis channel number, for example, if the sampling rates are high and if the sampling
rate ratio is not an integer number (e.g., if sr 1 = 22050 Hz and sr2 is 24000 Hz): In such a
case, the synthesis channel number might, for example, be selected as c2 = 2000).
In alternative embodiments, only the first sr1 or the second sr2 sampling rate is fed into the
synthesis channel number chooser 1010. In still further embodiments, neither the first sri
nor the second sr2 sampling rate is fed into the synthesis channel number chooser 1010,
and the synthesis channel number chooser 1010 then chooses a synthesis channel number
c2 independent of the sampling rates sr1, sr2.
The synthesis channel number chooser 1010 feeds the chosen synthesis channel number c2
into an analysis channel number calculator 1020. Furthermore, the first and second
sampling rate sri and sr2 are also fed into the analysis channel number calculator 1020. The
analysis channel number calculator calculates the number of analysis filter bank channels
c1 depending on the first and second sampling rate sri and sr2 and the synthesis channel
number c2 according to the formula:
Often, the situation may arise that the calculated number ci is not an integer number, but a
value being different from an integer number. However, the number of analysis filter bank
channels (as well as the number of synthesis filter bank channels) has to be an integer For
example, if a first sampling rate sr 1 is sr 1 = 22050 Hz, the second desired sampling rate sr2
is sr2 = 24000 Hz and the number of synthesis filter bank channels c2 has been chosen such
that c2 = 2000, then the calculated number of analysis channels c 1 . is c 1 = c2 · sr1/ sr2 = 2000
. 22050/24000 = 1837.5 analysis channels. Therefore, a decision has to be taken, whether
the analysis filter bank should comprise 1837 or 1838 channels.
Different rounding strategies may be applied:
According to one embodiment, a first rounding strategy is applied, according to which the
next lower integer value is chosen as analysis channel number, if the calculated value is not
an integer. E.g. a calculated value of 1837.4 or 1837.6 would be rounded to 1837.
According to another embodiment, a second rounding strategy is applied, according to
which the next higher integer value is chosen as analysis channel number, if the calculated
value is not an integer. E.g. a calculated value of 1837.4 or 1837,6 would be rounded to
1838,
According to a still further embodiment, arithmetic rounding is applied. E.g, a calculated
value of 1837,5 would be rounded to 1838 and a calculated value of 1837,4 would be
rounded to 1837.
However, as it is not possible in the "1837.5" example to apply the exact value of the
calculation as the number of analysis filter bank channels, not the desired second sampling
rate sr but a deviating actual second sampling rate sr2' will be obtained.
The controller of the embodiment of Fig. 16 comprises a sampling rate two calculator
1030, which calculates the actual second sampling rate sr based the first sampling rate
sri, the chosen number of synthesis filter bank channels c2 and the calculated number of
analysis filter bank channels ci according to the formula;
sr2' = c2 / c 1 . sr 1.
E.g. in the above described example, assuming that the first sampling rate sri is sr!=:22050
Hz, that the number of synthesis filter bank channels is c2 = 2000 and selecting the number
of analysis filter bank channels Cj to be 1838 this results in an actual second sampling rate
of:
instead of the desired 24000 Hz.
Applying an analysis filter bank having 1837 channels would result in an actual second
sampling rate of:
ST2 e2/ ci · sr1 - 2000/1837 · 22050 Hz = 24006.53 Hz instead of the desired 24000 Hz.
The actual second sampling rate sr2 ' of the audio signal being outputted from the synthesis
filter bank and the desired sampling rate sr2 are the fed into an error calculator 1040. The
error calculator calculates an actual error e' representing the difference between the desired
sampling rate sr? and the actual sampling rate sr2' according to the selected analysis and
synthesis filler bank channel setting.
in an embodiment, the actual error e' might be an absolute value of the difference between
the desired sampling rate sr2 and the actual sampling rate according to the formula:
e' = Isr2 - sr2' |.
In another embodiment, the actual error e' might be a relative value, e.g. calculated
according to the formula:
e' - I(sr2 - sr2') / sr2 .I
The error calculator then passes the actual error e' to an error comparator Ϊ 050. The error
comparator then compares the actual error e' with the tolerable error e. if the actual error e'
is within the bounds defined by the tolerable error, for example, if j e'j < je|, then the error
comparator 1050 instructs a channel number passer 1060 to pass the actual calculated
number of analysis filter bank channels to the analysis filter bank and the determined
number of the synthesis filter bank channels to the synthesis filter bank, respectively.
However, if the actual error e' is within the bounds defined by the tolerable error, for
example, if j e'j > je|, then the error comparator 1060 starts the determination process from
the beginning and instructs the synthesis channel number chooser 1010 to choose a
different synthesis channel number as number of synthesis filter bank channels.
Different embodiments may realize different strategies to choose a new synthesis channel
number. For example, in an embodiment, a synthesis channel number may be chosen
randomly in another embodiment, a higher channel number is chosen, e.g. a channel
number being twice the size of the synthesis channel number that was chosen by the
synthesis channel number chooser 1010, before. E.g. sr2 2 · sr2. For example, in the
above-mentioned example, the channel number sr2=2000 is replaced by sr2 := 2 . sr2 = 2 ·
2000 = 4000.
The process continues until a synthesis channel number with an acceptable actual error e'
has been found.
Fig. 17 illustrates a flow chart depicting a corresponding method, in step 1110, a synthesis
channel number c2 is chosen. In step 1120, the analysis channel number cj is calculated
based on the chosen synthesis channel number c2, the first sampling rate sri and the desired
sampling rate sr2. If necessary, rounding is performed to determine the analysis channel
number c\ . In step 1130, the actual second sampling rate is calculated based on the first
sampling rate srl s the chosen number of synthesis filter bank channels c2 and the calculated
number of analysis filter bank channels ci. Furthermore, in step 1140, an actual error e'
representing a difference between the actual second sampling rate sr2' and the desired
second sampling rate sr2 is calculated. In step 1150, the actual error e' is compared with a
defined tolerable error e. In case the error is tolerable, the process continues with step
1160: The chosen synthesis channel number is passed to the synthesis filter bank and the
calculated analysis channel number is passed to the analysis filter bank, respectively. If the
error is not tolerable, the process continues with step 1110, a new synthesis channel
number is chosen and the process is repeated until a suitable analysis and synthesis filter
bank channel number has been determined.
Fig. 18 illustrates a controller according to a further embodiment. The embodiment of Fig.
18 differs from the embodiment of Fig. 16 in that the synthesis channel number chooser
10 10 is replaced by an analysis channel number chooser 1210 and that the analysis channel
number calculator 1020 is replaced by a synthesis channel number calculator 1220. Instead
of choosing a synthesis channel number c2, the analysis channel number chooser 1210
chooses an analysis channel number c1. Then, the synthesis channel number calculator
1220 calculates a synthesis channel number c2 according to the formula c2 = c1 . sr2/ sr1.
The calculated synthesis channel number c2 is then passed to the sampling rate two
calculator 1230, which also receives the chosen analysis channel number cl s the first
sampling rate sri and the desired second sampling rate s¾. Apart from that, the sampling
rate two calculator 1230, the error calculator 1240, the error comparator 1250 and the
channel number passer 1260 correspond to the sampling rate two calculator 1030, the error
calculator 1040, the error comparator 1050 and the channel number passer 1060 of the
embodiment of Fig. 16, respectively.
Although some aspects 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,
The inventive decomposed signal can be stored on a digital storage medium or can be
transmitted on a transmission medium such as a wireless transmission medium or a wired
transmission medium such as the internet.
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 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.
Some embodiments according to the invention comprise a non-transitory 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, An apparatus for processing an audio signal, comprising:
a configurable first audio signal processor ( 110) for processing the audio signal (so)
in accordance with different configuration settings (conf) to obtain a processed
audio signal (si), wherein the apparatus is adapted so that different configuration
settings (conf) result in different sampling rates (SFJ) of the processed audio signal
( S1) ,
an analysis filter bank (120) having a first number (c 3) of analysis filter bank
channels,
a synthesis filter bank (130) having a second number (c2) of synthesis filter bank
channels,
a second audio processor (140) being adapted to receive and process an audio
signal (s2) having a predetermined sampling rate (sr2), and
a controller (150) for controlling the first number (cj) of analysis filter bank
channels or the second number ( c2 ) of synthesis filter bank channels in accordance
with a configuration setting (conf) provided to the configurable first audio signal
processor ( 110), so that an audio signal output (s2) of the synthesis filter bank (140)
has the predetermined sampling rate (sr2) or a sampling rate (sr2') being different
from the predetermined sampling rate (sr2) and being closer to the predetermined
sampling rate (sr2) than a sampling rate (srj) of an analysis filter bank input signal
(st).
2. An apparatus according to claim 1,
wherein the analysis filter bank (120) is adapted to transform the analysis filter bank
input signal (s1) being represented in a time-domain into a first time-frequency
domain audio signal having a plurality of first subband signals, wherein the number
of first subband signals is equal to the first number ( C1) of analysis filter bank
channels,
wherein the apparatus further comprises a signal adjuster (125) being adapted to
generate a second time-frequency domain audio signal having a plurality of second
subband signals from the first time-frequency domain audio signal based on the
configuration setting (conf), such that the number of second subband signals of the
second time-frequency domain audio signal is equal to the number (c2) of synthesis
filter bank channels, and wherein the number of second subband signals of the
second time-frequency domain audio signal is different from the number of subband
signals of the first time-frequency domain audio signal, and
wherein the synthesis filter bank (130) is adapted to transform the second timefrequency
domain audio signal into a time domain audio signal as the audio signal
output (s2) of the synthesis filter bank (130).
3 . An apparatus according to claim 2, wherein the signal adjuster (125) is adapted to
generate the second time-frequency domain audio signal by generating at least one
additional subband signal.
4 . An apparatus according to claim 3, wherein the signal adjuster (125) is adapted to
generate at least one addtional subband signal by conducting spectral band
replication to generate at least one additional subband signal.
An apparatus according to claim 3 or 4, wherein the signal adjuster (125) is adapted
5.
to generate a zero signal as additional subband signal
6 . An apparatus according to claim 1 or 2, wherein the analysis filter bank is a QMF
analysis filter bank and wherein the synthesis filter bank is a QMF synthesis filter
bank.
7 . An apparatus according to claim 1 or 2, wherein the analysis filter bank is an
MDCT analysis filter bank and wherein the synthesis filter bank is an MDCT
synthesis filter bank.
8 . An apparatus according to one of the preceding claims, wherein the apparatus
furthermore comprises an additional resampler (170) being adapted to receive a
synthesis filter bank output signal (sa) having a first synthesis sampling rate, and
wherein the additional resampler resamples the synthesis filter bank output signal to
receive a resampled output signal having a second synthesis sampling rate.
9. An apparatus according to one of claims 1 to 7, wherein the apparatus is adapted to
feed a synthesis filter hank output signal having a first synthesis sampling rate into
an analysis filter bank as an analysis filter bank input signal.
10. An apparatus according to one of the preceding claims, wherein the controller is
adapted to receive a configuration setting (conf) comprising an index number and
wherein the controller is adapted to determine the sampling rate (sr.) of the
processed audio signal (sa) or the predetermined sampling rate (sr2) based on the
index number and a lookup table,
11. An apparatus according to one of the preceding claims, wherein the controller is
adapted to determine the first number (ci) of analysis filter bank channels or the
second number (c2) of synthesis filter bank channels based on a tolerable error (e),
12. An apparatus according to claim 11, wherein the controller comprises an error
comparator ( 1050) for comparing the actual error (e5) with a tolerable error (e).
13. An apparatus for upmixing a surround signal comprising:
an analysis filter bank (910) for transforming a dowrmiixed time domain signal (si)
into a time-frequency domain to generate a plurality of downmixed subband signals
(Sn 3 S12? Sis),
at least two upmtx units (921, 922, 923) for upmixing the pluraliiy of subband
signals to obtain a plurality of surround subband signals,
at least two signal adjuster units (931, 932, 933, 934, 935) for adjusting the number
of surround subband signals, wherein the at least two signal adjuster units (93 1,
932, 933, 934, 935) are adapted to receive a first plurality of input surround
subband signals, wherein the at least two signal adjuster units (931, 932, 933, 934,
935) are adapted to output a second plurality of output surround subband signals,
and wherein the number of the first plurality of input surround subband signals and
the number of the second plurality of output surround subband signals is different,
a plurality of synthesis filter bank units (941, 942, 943, 944, 945) for transforming a
plurality of output surround subband signals from a time-frequency domain to a
time domain to obtain time domain surround output signals (s2 i , 822, s23, 824, S25),
and
a controller (950) being adapted to receive a configuration setting (conf) and being
adapted to control the number of channels of the analysis filter bank (910), to
control the number of channels of the synthesis filter bank units (941, 942, 943,
944, 945), to control the number of the first plurality of input surround subband
signals of the signal adjuster units (93 1, 932, 933, 934, 935), and to control the
number of the second plurality of output surround subband signals of the signal
adjuster units (931, 932, 933, 934, 935) based on the received configuration setting
(conf).
14. A method for processing an audio signal, comprising:
processing an audio signal in accordance with different configuration settings to
obtain a first processed audio signal, so that different configuration settings result in
different sampling rates of the first processed audio signal,
controlling a first number of analysis filter bank channels of an analysis filter bank
or a second number of synthesis filter bank channels of a synthesis filter bank in
accordance with a configuration setting, so that an audio signal output by the
synthesis filter bank has the predetermined sampling rate or a sampling rate being
different from the predetermined sampling rate and being closer to the
predetermined sampling rate than the sampling rate of an input signal into the
analysis filter bank, and
processing the audio signal output having the predetermined sampling rate.
15. Computer program for performing the method of claim 14, when the computer
program is executed by a computer or processor.