Apparatus and Method for Measuring a Plurality of Loudspeakers and Microphone
Array
Description
The present invention relates to acoustic measurements for loudspeakers arranged at
different positions in a listening area and, particularly, to an efficient measurement of a
high number of loudspeakers arranged in a three-dimensional configuration in the listening
area.
Fig. 2 illustrates a listening room at Fraunhofer IIS in Erlangen, Germany. This listening
room is necessary in order to perform listening tests. These listening tests are necessary in
order to evaluate audio coding schemes. In order to ensure comparable and reproducible
results of the listening tests, it is necessary to perform these tests in standardized listening
rooms, such as the listening room illustrated in Fig. 2. This listening room follows the
recommendation ITU-R BS 1116-1. In this room, the large number of 54 loudspeakers is
mounted as a three-dimensional loudspeaker set-up. The loudspeakers are mounted on a
two-layered circular truss suspended from the ceiling and on a rail system on the wall. The
large number of loudspeakers provides great flexibility, which is necessary, both for
academic research and to study current and future sound formats.
With such a large number of loudspeakers, verifying that they are working correctly and
that they are properly connected is a tedious and cumbersome task. Typically, each
loudspeaker has individual settings at the loudspeaker box. Additionally, an audio matrix
exists, which allows switching certain audio signals to certain loudspeakers. In addition, it
cannot be guaranteed that all loudspeakers, apart from the speakers, which are fixedly
attached to a certain support, are at their correct positions. In particular, the loudspeakers
standing on the floor in Fig. 2 can be shifted back and forth and to the left and right and,
therefore, it cannot be guaranteed that, at the beginning of a listening test, all speakers are
at the position at which they should be, all speakers have their individual settings as they
should have and that the audio matrix is set to a certain state in order to correctly distribute
loudspeaker signals to the loudspeakers. Apart from the fact that such listening rooms are
used by a plurality of research groups, electrical and mechanical failures can occur from
time to time.
In particular, the following exemplary problems can occur. These are:
Loudspeakers not switched on or not connected
Signal routed to the wrong loudspeaker, signal cable connected to the
wrong loudspeaker
Level of one loudspeaker wrongly adjusted in the audio routing system
or at the loudspeaker
Wrongly set equalizer in the audio routing system or at the loudspeaker
Damage of a single driver in a multi-way loudspeaker
Loudspeaker is wrongly placed, oriented or an object is obstructing the
acoustic pathway.
Normally, in order to manually evaluate the functionality of the loudspeaker set-up in the
listening area, a great amount of time is necessary. This time is required for manually
verifying the position and orientation of each loudspeaker. Additionally, each loudspeaker
has to be manually inspected in order to find out the correct loudspeaker settings. In order
to verify the electrical functionality of the signal routing on the one hand and the individual
speakers on the other hand, a highly experienced person is necessary to perform a listening
test where, typically, each loudspeaker is excited with the test signal and the experienced
listener then evaluates, based on his knowledge, whether this loudspeaker is correct or not.
It is clear that this procedure is expensive due to the fact that a highly experienced person
is necessary. Additionally, this procedure is tedious due to the fact that the inspection of all
loudspeakers will typically reveal that most, or even all, loudspeakers are correctly
oriented and correctly set, but on the other hand, one cannot dispense with this procedure,
since a single or several faults, which are not discovered, can destroy the significance of a
listening test. Finally, even though an experienced person conducts the functionality
analysis of the listening room, errors are, nevertheless, not excluded.
It is the object of the present invention to provide an improved procedure for verifying the
functionality of a plurality of loudspeakers arranged at different positions in a listening
area.
This object is achieved by an apparatus for measuring a plurality of loudspeakers in
accordance with claim 1, a method of measuring a plurality of loudspeakers in accordance
with claim 11, a computer program in accordance with claim 12 or a microphone array in
accordance with claim 13.
The present invention is based on the finding that the efficiency and the accuracy of
listening tests can be highly improved by adapting the verification of the functionality of
the loudspeakers arranged in the listening space using an electric apparatus. This apparatus
comprises a test signal generator for generating a test signal for the loudspeakers, a
microphone device for picking up a plurality of individual microphone signals, a controller
for controlling emissions of the loudspeaker signals and the handling of the sound signal
recorded by the microphone device, so that a set of sound signals recorded by the
microphone device is associated with each loudspeaker, and an evaluator for evaluating the
set of sound signals for each loudspeaker to determine at least one loudspeaker
characteristic for each loudspeaker and for indicating a loudspeaker state using the at least
one loudspeaker characteristic.
The invention is advantageous in that it allows to perform the verification of loudspeakers
positioned in a listening space by an untrained person, since the evaluator will indicate an
OK/non-OK state and the untrained person can individually examine the non-OK
loudspeaker and can rely on the loudspeakers, which have been indicated to be in a
functional state.
Additionally, the invention provides great flexibility in that individually selected
loudspeaker characteristics and, preferably, several loudspeaker characteristics can be used
and calculated in addition, so that a complete picture of the loudspeaker state for the
individual loudspeakers can be gathered. This is done by providing a test signal to each
loudspeaker, preferably in a sequential way and by recording the loudspeaker signal
preferably using a microphone array. Hence, the direction of arrival of the signal can be
calculated, so that the position of the loudspeaker in the room, even when the loudspeakers
are arranged in a three-dimensional scheme, can be calculated in an automatic way.
Specifically, the latter feature cannot be fulfilled even by an experienced person typically
in view of the high accuracy, which is provided by a preferred inventive system.
In a preferred embodiment, a multi-loudspeaker test system can accurately determine the
position within a tolerance of ± 3° for the elevation angle and the azimuth angle. The
distance accuracy is ± 4 cm and the magnitude response of each loudspeaker can be
recorded in an accuracy of ± ldB of each individual loudspeaker in the listening room.
Preferably, the system compares each measurement to a reference and can so identify the
loudspeakers, which are operating outside the tolerance.
Additionally, due to reasonable measurement times, which are as low as 10 s per
loudspeaker including processing, the inventive system is applicable in practice even when
a large number of loudspeakers have to be measured. In addition, the orientation of the
loudspeakers is not limited to any certain configuration, but the measurement concept is
applicable for each and every loudspeaker arrangement in an arbitrary three-dimensional
scheme.
Preferred embodiments of the present invention will subsequently be discussed with
reference to the Figs., in which:
Fig. 1 illustrates a block diagram of an apparatus for measuring a plurality of
loudspeakers;
Fig. 2 illustrates an exemplary listening test room with a set-up of 9 main
loudspeakers, 2 sub woofers and 43 loudspeakers on the walls and the two
circular trusses on different heights;
Fig. 3 illustrates a preferred embodiment of a three-dimensional microphone array;
Fig. 4a illustrates a schematic for illustrating steps for determining the direction of
arrival of the sound using the DirAC procedure;
Fig. 4b illustrates equations for calculating particle velocity signals in different
directions using microphones from the microphone array in Fig. 3;
Fig. 4c illustrates a calculation of an omnidirectional sound signal for a B-format,
which is performed when the central microphone is not present;
Fig. 4d illustrates steps for performing a three-dimensional localization algorithm;
Fig. 4e illustrates a real spatial power density for a loudspeaker;
Fig. 5 illustrates a schematic of a hardware set of loudspeakers and microphones;
Fig. 6a illustrates a measurement sequence for reference;
Fig. 6b illustrates a measurement sequence for testing;
Fig. 6c illustrates an exemplary measurement output in the form of a magnitude
response where, in a certain frequency range, the tolerances are not fulfilled;
Fig. 7 illustrates a preferred implementation for determining several loudspeaker
characteristics;
Fig. 8 illustrates an exemplary pulse response and a window length for performing
the direction of arrival determination; and
Fig. 9 illustrates the relations of the lengths of portions of impulse response(s)
required for measuring the distance, the direction of arrival and the impulse
response/transfer function of a loudspeaker.
Fig. 1 illustrates an apparatus for measuring a plurality of loudspeakers arranged at
different positions in a listening space. The apparatus comprises a test signal generator 10
for generating a test signal for a loudspeaker. Exemplarily, N loudspeakers are connected
to the test signal generator at loudspeaker outputs 10a, . . ., 10b.
The apparatus additionally comprises a microphone device 1 . The microphone device 12
may be implemented as a microphone array having a plurality of individual microphones,
or may be implemented as a microphone, which can be sequentially moved between
different positions, where a sequential response by the loudspeaker to sequentially applied
test signals is measured for the microphone device is configured for receiving sound
signals in response to one or more loudspeaker signals emitted by a loudspeaker of the
plurality of loudspeakers in response to one or more test signals.
Additionally, a controller 14 is provided for controlling emissions of the loudspeaker
signals by the plurality of loudspeakers and for handling the sound signals received by the
microphone device so that a set of sound signals recorded by the microphone device is
associated with each loudspeaker of the plurality of loudspeakers in response to one or
more test signals. The controller 14 is connected to the microphone device via signal lines
13a, 13b, 13c. When the microphone device only has a single microphone movable to
different positions in a sequential way, a single line 13a would be sufficient.
The apparatus for measuring additionally comprises an evaluator 16 for evaluating the set
of sound signals for each loudspeaker to determine at least one loudspeaker characteristic
for each loudspeaker and for indicating a loudspeaker state using the at least one
loudspeaker characteristic. The evaluator is connected to the controller via a connection
line 17, which can be a single direction connection from the controller to the evaluator, or
which can b e a two-way connection when the evaluator is implemented to provide
information to the controller. Thus, the evaluator provides a state indication for each
loudspeaker, i.e. whether this loudspeaker is a functional loudspeaker or is a defective
loudspeaker.
Preferably, the controller 14 is configured for performing an automatic measurement in
which a certain sequence is applied for each loudspeaker. Specifically, the controller
controls the test signal generator to output a test signal. At the same time, the controller
records signals picked up the microphone device and the circuits connected to the
microphone device, when a measurement cycle is started. When the measurement of the
loudspeaker test signal is completed, the sound signals received by each of the
microphones are then handled by the controller and are e.g. stored by the controller in
association with the specific loudspeaker, which has emitted the test signal or, more
accurately, which was the device under test. As stated before, it is to be verified whether
the specific loudspeaker, which has received the test signal is, in fact, the actual
loudspeaker, which finally has emitted a sound signal corresponding to the test signal. This
is verified by calculating the distance or direction of arrival of the sound emitted by the
loudspeaker in response to the test signal preferably using the directional microphone
array.
Alternatively, the controller can perform a measurement of several or all loudspeakers
concurrently. To this end, the test signal generator is configured for generating different
test signals for different loudspeakers. Preferably, the test signals are at least partly
mutually orthogonal to each other. This orthogonality can include different nonoverlapping
frequency bands in a frequency multiplex or different codes in a code
multiplex or other such implementations. The evaluator is configured for separating the
different test signals for the different loudspeakers such as by associating a certain
frequency band to a certain loudspeaker or a certain code to a certain loudspeaker in
analogy to the sequential implementation, in which a certain time slot is associated to a
certain loudspeaker.
Thus, the controller automatically controls the test signal generator and handles the signals
picked up by the microphone device to generate the test signals e.g. in a sequential manner
and to receive the sound signals in a sequential manner so that the set of sound signals is
associated with the specific loudspeaker, which has emitted the loudspeaker test signal
immediately before a reception of the set of sound signals by the microphone array.
A schematic of the complete system including the audio routing system, loudspeakers,
digital/analog converter, analog/digital converters and the three-dimensional microphone
array is presented in Fig. 5 . Specifically, Fig. 5 illustrates an audio routing system 50, a
digital/analog converter for digital/analog converting a test signal input into a loudspeaker
where the digital/ analog converter is indicated at 5 1. Additionally, an analog/digital
converter 52 is provided, which is connected to analog outputs of individual microphones
arranged at the three-dimensional microphone array 12. Individual loudspeakers are
indicated at 54a, . . ., 54b. The system may comprise a remote control 55 which has the
functionality for controlling the audio routing system 50 and a connected computer 56 for
the measurement system. The individual connections in the preferred embodiment are
indicated at Fig. 5 where "MADI" stands for multi-channel audio/digital interface, and
"ADAT" stands for Ale sis-digital-audio-tape (optical cable format). The other
abbreviations are known to those skilled in the art. A test signal generator 10, the controller
14 and the evaluator 16 of Fig. 1 are preferably included in the computer 56 of Fig. 5 or
can also be included in the remote control processor 55 in Fig. 5.
Preferably, the measurement concept is performed on the computer, which is normally
feeding the loudspeakers and controls. Therefore, the complete electrical and acoustical
signal processing chain from the computer over the audio routing system, the loudspeakers
until the microphone device at the listening position is measured. This is preferred in order
to capture all possible errors, which can occur in such a signal processing chain. The single
connection 57 from the digital/analog converter 5 1 to the analog/digital converter 52 is
used to measure the acoustical delay between the loudspeakers and the microphone device
and can be used for providing the reference signal X illustrated at Fig. 7 to the evaluator 16
of Fig. 1, so that a transfer function or, alternatively, an impulse response from a selected
loudspeaker to each microphone can be calculated by convolution as known in the art.
Specifically, Fig. 7 illustrates a step 70 performed by the apparatus illustrated in Fig. 1 in
which the microphone signal Y is measured, and the reference signal X is measured, which
is done by using the short-circuit connection 57 in Fig. 5. Subsequently, in the step 71, a
transfer function H can be calculated in the frequency domain by division of frequencydomain
values or an impulse response h(t) can be calculated in the time domain using
convolution. The transfer function H(f) is already a loudspeaker characteristic, but other
loudspeaker characteristics as exemplarily illustrated in Fig. 7 can be calculated as well.
These other characteristics are, for example, the time domain impulse response h(t), which
can be calculated by performing an inverse FFT of the transfer function. Alternatively, the
amplitude response, which is the magnitude of the complex transfer function, can be
calculated as well. Additionally, the phase as a function of frequency can be calculated or
the group delay t, which is the first derivation of the phase with respect to frequency. A
different loudspeaker characteristic is the energy time curve, etc., which indicates the
energy distribution of the impulse response. An additional important characteristic is the
distance between the loudspeaker and a microphone and a direction of arrival of the sound
signal at the microphone is an additional important loudspeaker characteristic, which is
calculated using the DirAC algorithm, as will be discussed later on.
The Fig. 1 system presents an automatic multi-loudspeaker test system, which, by
measuring each loudspeaker's position and magnitude response, verifies the occurrence of
the above-described variety of problems. All these errors are detectable by post-processing
steps carried out by the evaluator 16 of Fig. 1. To this end, it is preferred that the evaluator
calculates room impulse responses from the microphone signals which have been recorded
with each individual pressure microphone from the three-dimensional microphone array
illustrated in Fig. 3 .
Preferably, a single logarithmic sine sweep is used as a test signal, where this test signal is
individually played by each speaker under test. This logarithmic sine sweep is generated by
the test signal generator 10 of Fig. 1 and is preferably equal for each allowed speaker. The
use of this single test signal to check for all errors is particularly advantageous as it
significantly reduces the total test time to about 10 s per loudspeaker including processing.
Preferably, impulse response measurements are formed as discussed in the context of Fig.
7 where a logarithmic sine sweep is used as the test signal is optimal in practical acoustic
measurements with respect to good signal-to-noise ratio, also for low frequencies, not too
much energy in the high frequencies (no tweeter damaging signal), a good crest factor and
a non-critical behavior regarding small non-linearities.
Alternatively, maximum length sequences (MLS) could also be used, but the logarithmic
sine sweep is preferable due to the crest factor and the behavior against non-linearities.
Additionally, a large amount of energy in the high frequencies might damage the
loudspeakers, which is also an advantage for the logarithmic since sweep, since this signal
has less energy in the high frequencies.
Figs. 4a to 4e will subsequently be discussed to show a preferred implementation of the
direction of arrival estimation, although other direction of arrival algorithms apart from
DirAC can be used as well. Fig. 4a schematically illustrates the microphone array 12
having 7 microphones, a processing block 40 and a DirAC block 42. Specifically, block 40
performs short-time Fourier analysis of each microphone signal and, subsequently,
performs the conversion of these preferably 7 microphone signals into the B-format having
an omnidirectional signal W and having three individual particle velocity signals X, Y, Z
for the three spatial directions X, Y, Z, which are orthogonal to each other.
Directional audio coding is an efficient technique to capture and reproduce spatial sound
on the basis of a downmix signal and side information, i.e. direction of arrival (DOA) and
diffuseness of the sound field. DirAC operates in the discrete short-time Fourier transform
(STFT) domain, which provides a time-variant spectral representation of the signals. Fig.
4a illustrates the main steps for obtaining the DOA with DirAC analysis. Generally, DirAC
requires B-format signals as input, which consists of sound pressure and particle velocity
vector measured in one point in space. It is possible from this information to compute the
active intensity vector. This vector describes direction and magnitude of the net flow of
energy characterizing the sound field in the measurement position. The DOA of a sound is
derived from the intensity vector by taking the opposite to its direction and it is expressed,
for example, by azimuth and elevation in a standard spherical coordinate system.
Naturally, other coordinate systems can be applied as well. The required B-format signal is
obtained using a three-dimensional microphone array consisting of 7 microphones
illustrated in Fig. 3 . The pressure signal for the DirAC processing is captured by the central
microphone 7 in Fig. 3, whereas the components of the particle velocity vector are
estimated from the pressure difference between opposite sensors along the three Cartesian
axes. Specifically, Fig. 4b illustrates the equations for calculating the sound velocity vector
U(k,n) having the three components Ux,Uy and Uz.
Exemplarily, the variable Pi stands for the pressure signal of microphone Rl of Fig. 3 and,
for example, P 3 stands for the pressure signal of microphone R3 in Fig. 3. Analogously, the
other indices in Fig. 4b correspond to the corresponding numbers in Fig. 3. k denotes a
frequency index and n denotes a time block index. All quantities are measured in the same
point in space. The particle velocity vector is measured along two or more dimensions. For
the sound pressure P(k,n) of the B-format signal, the output of the center microphone R7 is
used. Alternatively, if no center microphone is available, P(k,n) can be estimated by
combining the outputs of the available sensors, as illustrated in Fig. 4c. It is to be noted
that the same equations also hold for the two-dimensional and one-dimensional case. In
these cases, the velocity components in Fig. 4b are only calculated for the considered
dimensions. It is to be further noted that the B-format signal can be computed in time
domain in exactly the same way. In this case, all frequency domain signals are substituted
by the corresponding time-domain signals. Another possibility to determine a B-format
signal with microphone arrays is to use directional sensors to obtain the particle velocity
components. In fact, each particle velocity component can be measured directly with a bidirectional
microphone (a so-called figure-of-eight microphone). In this case, each pair of
opposite sensors in Fig. 3 is replaced by a bi-directional sensor pointing along the
considered axis. The outputs of the bi-directional sensors correspond directly to the desired
velocity components.
Fig. 4d illustrates a sequence of steps for performing the DOA in the form of azimuth on
the one hand and elevation on the other hand. In a first step, an impulse response
measurement for calculating impulse responses for each of the microphones is performed
in step 43. A windowing at the maximum of each impulse response is then performed, as
exemplarily illustrated in Fig. 8 where the maximum is indicated at 80. The windowed
samples are then transformed into a frequency domain at block 45 in Fig. 4d. In the
frequency domain, the DirAC algorithm is performed for calculating the DOA in each
frequency bin of, for example, 20 frequency bins or even more frequency bins. Preferably,
only a short window length of, for example, only 512 samples is performed, as illustrated
at an FFT 512 in Fig. 8 so that only the direct sound at maximum 80 until the early
reflections, but preferably excluding the early reflections, is used. This procedure provides
a good DOA result, since only sound from an individual position without any
reverberations is used.
As indicated at 46, the so-called spatial power density (SPD) is then calculated, which
expresses, for each determined DOA, the measured sound energy.
Fig. 4e illustrates a measured SPD for a loudspeaker position with elevation and azimuth
equal to 0°. The SPD shows that most of the measured energy is concentrated around
angles, which correspond to the loudspeaker position. In ideal scenarios, i.e. where no
microphone noise is present, it would be sufficient to determine the maximum of the SPD
in order to obtain the loudspeaker position. However, in a practical application, the
maximum of the SPD does not necessarily correspond to the correct loudspeaker position
due to measurement inaccuracies. Therefore, it is simulated, for each DOA, a theoretical
SPD assuming zero mean white Gaussian microphone noise. By comparing the theoretical
SPDs with the measured SPD (exemplarily illustrated in Fig. 4e), the best fitting
theoretical SPD is determined whose corresponding DOA then represents the most likely
loudspeaker position.
Preferably, in a non-reverberant environment, the SPD is calculated by the downmix audio
signal power for the time/frequency bins having a certain azimuth/elevation. When this
procedure is performed in the reverberating environment or when early reflections are used
as well, the long-term spatial power density is calculated from the downmix audio signal
power for the time/frequency bins, for which a diffuseness obtained by the DirAC
algorithm is below a specific threshold. This procedure is described in detail in AES
convention paper 7853, October 9, 2009 "Localization of Sound Sources in Reverberant
Environments based on Directional Audio Coding Parameters", O. Thiergart, et al.
Fig. 3 illustrates a microphone array having three pairs of microphones. The first pair are
microphones Rl and R3 in a first horizontal axis. The second pair of microphones consists
of microphones R2 and R4 in a second horizontal axis. The third pair of microphones
consists of microphones R5 and R6 representing the vertical axis, which is orthogonal to
the two orthogonal horizontal axes.
Additionally, the microphone array consists of a mechanical support for supporting each
pair of microphones at one corresponding spatial axis of the three orthogonal spatial axes.
In addition, the microphone array comprises a laser 30 for registration of the microphone
array in the listening space, the laser being fixedly connected to the mechanical support so
that a laser ray is parallel or coincident with one of the horizontal axes.
The microphone array preferably additionally comprises a seventh microphone R7 placed
at a position in which the three axes intersect each other. As illustrated in Fig. 3, the
mechanical support comprises the first mechanical axis 31 and the second horizontal axis
32 and a third vertical axis 33. The third horizontal axis 33 is placed in the center with
respect to a "virtual" vertical axis formed by a connection between microphone R5 and
microphone R6. The third mechanical axis 33 is fixed to an upper horizontal rod 34a and a
lower horizontal rod 34b where the rods are parallel to the horizontal axes 3 1 and 32.
Preferably, the third axis 33 is fixed to one of the horizontal axes and, particularly, fixed to
the horizontal axis 32 at the connection point 35. The connection point 35 is placed
between the reception for the seventh microphone R7 and a neighboring microphone, such
as microphone R2 of one pair of the three pairs of microphones. Preferably, the distance
between the microphones of each pair of microphones is between 4 cm and 10 cm or even
more preferably between 5 cm and 8 cm and, most preferably, at 6.6 cm. This distance can
be equal for each of the three pairs, but this is not a necessary condition. Rather small
microphones Rl to R7 are used and thin mounting is necessary for ensuring acoustical
transparency. To provide reproducibility of the results, precise positioning of the single
microphones and of the whole array is required. The latter requirement is fulfilled by
employing the fixed cross-laser pointer 30, whereas the former requirement is achieved
with a stable mounting. To obtain accurate room impulse response measurements,
microphones characterized by a flat magnitude response are preferred. Moreover, the
magnitude responses of different microphones should be matched and should not change
significantly in time to provide reproducibility of the results. The microphones deployed in
the array are high quality omnidirectional microphones DPA 4060. Such a microphone has
an equivalent noise level A-weighted of typically 26 dBA re. 20 mPa and a dynamic range
of 97 dB. The frequency range between 20 Hz and 20 kHz is in between 2 dB from the
nominal curve. The mounting is realized in brass, which ensures the necessary mechanical
stiffness and, at the same time, the absence of scattering. The usage of omnidirectional
pressure microphones in the array in Fig. 3 compared to bi-directional figure-of-eight
microphones is preferable in that individual omnidirectional microphones are considerably
cheaper compared to expensive by-directional microphones.
The measurement system is particularly indicated to detect changes in the system with
respect to a reference condition. Therefore, a reference measurement is first carried out, as
illustrated in Fig. 6a. The procedure in Fig. 6a and in Fig. 6b is performed by the controller
14 illustrated in Fig. 1. Fig. 6a illustrates a measurement for each loudspeaker at 60 where
the sinus sweep is played back and the seven microphone signals are recorded at 61. A
pause 62 is then conducted and, subsequently, the measurements are analyzed 63 and
saved 64. The reference measurements are performed subsequent to a manual verification
in that, for the reference measurements, all loudspeakers are correctly adjusted and at the
correct position. These reference measurements must be performed only a single time and
can be used again and again.
The test measurements should, preferably, be performed before each listening test. The
complete sequence of test measurements is presented in Fig. 6b. In a step 65, control
settings are read. Next, in step 66, each loudspeaker is measured by playing back the sinus
sweep and by recording the seven microphone signals and the subsequent pause. After that,
in step 67, a measurement analysis is performed and in step 68, the results are compared
with the reference measurement. Next, in step 69, it is determined whether the measured
results are inside the tolerance range or not. In a step 73, a visional presentation of results
can be performed and in step 74, the results can be saved.
Fig. 6c illustrates an example for visual presentation of the results in accordance with step
73 of Fig. 6b. The tolerance check is realized by setting an upper and lower limit around
the reference measurement. The limits are defined as parameters at the beginning of the
measurement. Fig. 6c visualizes the measurement output regarding the magnitude
response. Curve 3 is the upper limit of the reference measurement and curve 5 is the lower
limit. Curve 4 is the current measurement. In this example, a discrepancy in the midrange
frequency is shown, which is visualized in the graphical user interface (GUI) by red
markers at 75. This violation of the lower limit is also shown in field 2. In a similar
fashion, the results for azimuth, elevation, distance and polarity are presented in the
graphical user interface.
Fig. 9 will subsequently be described in order to illustrate the three preferred main
loudspeaker characteristics, which are calculated for each loudspeaker in the measuring of
a plurality of loudspeakers. The first loudspeaker characteristic is the distance. The
distance is calculated using the microphone signal generated by microphone R7. To this
end, the controller 14 of Fig. 1 controls the measurement of the reference signal X and the
microphone signal Y of the center microphone R7. Next, the transfer function of the
microphone signal R7 is calculated, as outlined in step 71. In this calculation, a search for
the maximum, such as 80 in Fig. 8 of the impulse response calculated in step 7 1 is
performed. Afterwards, this time at which the maximum 80 occurs is multiplied by the
sound velocity v in order to obtain the distance between the corresponding loudspeaker and
the microphone array.
To this end, only a short portion of the impulse response obtained from the signal of
microphone R7 is required, which is indicated as a "first length" in Fig. 9. This first length
only extends from 0 to the time of the maximum 80 and including this maximum, but not
including any early reflections or diffuse reverberations. Alternatively, any other
synchronization can be performed between the test signal and the response from the
microphone, but using a first small portion of the impulse response calculated from the
microphone signal of microphone R7 is preferred due to efficiency and accuracy.
Next, for the DOA measurements, the impulse responses for all seven microphones are
calculated, but only a second length of the impulse response, which is longer than the first
length, is used and this second length preferably extends only up to the early reflections
and, preferably, do not include the early reflections. Alternatively, the early reflections are
included in the second length in an attenuated state determined by a side portion of a
window function, as e.g. illustrated in Fig. 8 by window shape 81. The side portion has
window coefficients smaller than 0.5 or even smaller than 0.3 compared to window
coefficients in the mid portion of the window, which approach 1.0. The impulse responses
for the individual microphones Rl to R7 are preferably calculated, as indicated by steps 70,
71.
Preferably a window is applied to each impulse response or a microphone signal different
from the impulse response, wherein a center of the window or a point of the window
within 50 percents of the window length centered around the center of the window is
placed at the maximum in each impulse response or a time in the microphone signal
corresponding to the maximum to obtain a windowed frame for each sound signal
The third characteristic for each loudspeaker is calculated using the microphone signal of
microphone 5, since this microphone is not influenced too much by the mechanical
support of the microphone array illustrated in Fig. 3. The third length of the impulse
response is longer than the second length and, preferably, includes not only the early
reflections, but also the diffuse reflections and may extend over a considerable amount of
time, such as 0.2 ms in order to have all reflections in the listening space. Naturally, when
the room is a quite non-reverberant room, then the impulse response of microphone R5
will be close to 0 quite earlier. In any case, however, it is preferred to use a short length of
the impulse response for a distance measurement, to use the medium second length for the
DOA measurements and to use a long length for measuring the loudspeaker impulse
response/transfer function, as illustrated at the bottom of Fig. 9 .
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.
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 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.
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Claims
1. Apparatus for measuring a plurality of loudspeakers arranged at different positions,
comprising:
a test signal generator (10) for generating a test signal for a loudspeaker;
a microphone device (12) being configured for receiving a plurality of different
sound signals in response to one or more loudspeaker signals emitted by a
loudspeaker of the plurality of loudspeakers in response to the test signal;
a controller (14) for controlling emissions of the loudspeaker signals by the
plurality of loudspeakers and for handling the plurality of different sound signals so
that a set of sound signals recorded by the microphone device is associated with
each loudspeaker of the plurality of loudspeakers in response to the test signal; and
an evaluator (16) for evaluating the set of sound signals for each loudspeaker to
determine at least one loudspeaker characteristic for each loudspeaker and for
indicating a loudspeaker state using the at least one loudspeaker characteristic for
the loudspeaker.
2. Apparatus in accordance with claim 1, in which the controller (14) is configured for
automatically controlling the test signal generator (10) and the microphone device
(12) to generate the test signals in a sequential manner and to receive the sound
signals in a sequential manner so that the set of sound signals is associated with the
specific loudspeaker, which has emitted the loudspeaker test signal immediately
before a reception of the set of sound signals, or .
in which the controller (14) is configured for automatically controlling the test
signal generator (10) and the microphone device (12) to generate the test signals in
a parallel manner and to demultiplex the sound signals so that the set of sound
signals is associated with the specific loudspeaker, which is associated to a certain
frequency band of the set of sound signals or which is associated to a certain code
sequence in a code multiplexed test signal.
3. Apparatus in accordance with claim 1 or 2 , in which the evaluator (16) i s
configured for calculating a distance between the loudspeaker position for a
loudspeaker and the microphone device by using a time delay value of a maximum
of an impulse response of a sound signal between the loudspeaker and the
microphone device and by using the sound velocity in air.
Apparatus in accordance with one of the preceding claims, in which the controller
(14) is configured for performing a reference measurement using the test signal (70)
in which an analog output of a digital/analog converter (51) to a loudspeaker and an
analog input of an analog/digital converter (52) to which the microphone device are
connected is directly connected to determine reference measurement data; and
in which the evaluator (16) is configured to determine a transfer function or an
impulse response for a selected microphone of the plurality of microphones using
the reference measurement data to determine an impulse response or a transfer
function for the loudspeaker as the loudspeaker characteristic.
Apparatus according to one of the preceding claims,
in which the evaluator (16) is configured for calculating a direction of arrival for
sound emitted by a loudspeaker using the set of sound signals, wherein the
evaluator is adapted for
transforming (40) the set of test signals into B-format signals having an
omnidirectional signal (W) and at least two particle velocity signals (X, Y, Z) for at
least two orthogonal directions in space;
calculating, for each frequency bin of a plurality of frequency bins, a direction of
arrival result; and
determining (46, 47) the direction of arrival for the sound emitted by the
loudspeaker using the direction of arrival results for the plurality of frequency bins.
Apparatus in accordance with claim 5, in which the evaluator (16) is configured for
calculating an impulse response for each microphone,
for searching a maximum in each impulse response;
for applying a window to each impulse response or a microphone signal different
from the impulse response, wherein a center of the window or a point of the
window within 50 percents of the window length centered around the center of the
window is placed at the maximum in each impulse response or a time in the
microphone signal corresponding to the maximum to obtain a windowed frame for
each sound signal; and
for converting each frame from the time domain to a spectral domain.
Apparatus according to one of the preceding claims, in which the microphone
device comprises a microphone array comprising three pairs of microphones
arranged on three spatial axes;
wherein an omnidirectional pressure signal is derived by the evaluator by using the
signals received by the three pairs or using a further microphone arranged at a point
in which the three spatial axes intersect each other.
Apparatus in accordance with claim 7,
in which the evaluator (16) is configured for
calculating a distance between the microphone array and a loudspeaker using the
omnidirectional pressure signal, wherein the omnidirectional pressure signal has a
first length in samples, the first length extending to a maximum of the
omnidirectional pressure signal;
calculating an impulse response or transfer function of the loudspeaker using a
microphone signal from an individual microphone of the three pairs, the
microphone signal having a third length in samples, the third length having at least
a direct sound maximum and early reflections, the third length being longer than the
first length; and
calculating a direction of arrival of the sound from the loudspeaker using signals
from all microphones, the signals having a second length in samples being longer
than the first length and shorter than the third length, the second length including
values up to an early reflection so that the early reflections are not included in the
second length or are included in the second length in an attenuated state determined
by a side portion of a window function.
9. Apparatus in accordance with claim 5, in which the evaluator (16) is configured for
determining the direction of arrival by calculating a real spatial power density
having a value for each elevation angle and for each azimuth angle, and
for providing a plurality of ideal spatial power densities assuming zero mean white
Gaussian microphone noise for different elevation angles and azimuth angles, and
selecting (47) the elevation angle and azimuth angle belonging to the ideal spatial
power density, which has a best fit to the real spatial power density.
10. Apparatus in accordance with one of the preceding claims, in which the evaluator is
configured for comparing the at least one loudspeaker characteristic to an expected
loudspeaker characteristic and to indicate a loudspeaker having the at least one
loudspeaker characteristic equal to the expected loudspeaker characteristic as a
functional loudspeaker and to indicate a loudspeaker not having the at least one
loudspeaker characteristic equal to the expected loudspeaker characteristic as a non
functional loudspeaker.
11. Method of measuring a plurality of loudspeakers arranged at different positions in a
listening space, comprising:
generating (10) a test signal for a loudspeaker;
receiving a plurality of different sound signals by a microphone device in response
to one or more loudspeaker signals emitted by a loudspeaker of the plurality of
loudspeakers in response to the test signal;
controlling (14) emissions of the loudspeaker signals by the plurality of
loudspeakers and handling the plurality of different sound signals so that a set of
sound signals recorded by the microphone device is associated with each
loudspeaker of the plurality of loudspeakers in response to the test signal; and
evaluating (16) the set of sound signals for each loudspeaker to determine at least
one loudspeaker characteristic for each loudspeaker and indicating a loudspeaker
state using the at least one loudspeaker characteristic for the loudspeaker.
12. Computer program for performing a computer program implementing the method
of claim 11, when running on a processor.
13. Microphone array comprising:
three pairs of microphones (Rl, R2, R3, R4, R5, R6); and
a mechanical support for supporting each pair of microphones at one spatial axis of
three orthogonal spatial axes, the three spatial axes having two horizontal axes and
one vertical axis.
14. Microphone array in accordance with claim 13, further comprising:
a laser (30) for registration of the microphone array in a listening room, the laser
being fixedly connected to the mechanical support so that a laser ray is parallel or
coincident with one of the horizontal axes (31, 32).
15. Microphone array in accordance with claim 13 or 14, further comprising a seventh
microphone (R7) placed at the position in which the three axes intersect each other,
wherein the mechanical support comprises a first horizontal mechanical axis (31)
and a second horizontal mechanical axis (32) and a third vertical mechanical axis
(33) being placed off-center with respect to a virtual vertical axis intersecting a
cross-point of the two horizontal mechanical axes (31, 32),
wherein the third axis (33) is fixed to an upper horizontal rod (34a) and a lower
horizontal rod (34b), the rods (34a, 34b) being parallel to the horizontal axis, and
wherein the third axis (33) is fixed to one of the horizontal axes between a place for
the seventh microphone (R7) and a neighboring microphone (R2) of one pair of the
three pairs of microphones at a connection place (35).
16. Microphone array in accordance with claim 14 or 15,
in which a distance between the microphones of each pair of microphones is
between 5 cm and 8 cm.
17. Microphone array of one of claims 13 to 16, in which all microphones are pressure
microphones fixed at the mechanical support so that the microphones are oriented
in the same direction.