EP1125274B1 - Digital audio signal processing apparatus comprising a delay line - Google Patents
Digital audio signal processing apparatus comprising a delay line Download PDFInfo
- Publication number
- EP1125274B1 EP1125274B1 EP00960453A EP00960453A EP1125274B1 EP 1125274 B1 EP1125274 B1 EP 1125274B1 EP 00960453 A EP00960453 A EP 00960453A EP 00960453 A EP00960453 A EP 00960453A EP 1125274 B1 EP1125274 B1 EP 1125274B1
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- European Patent Office
- Prior art keywords
- filter
- signal
- filters
- output
- processor
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H1/00—Details of electrophonic musical instruments
- G10H1/02—Means for controlling the tone frequencies, e.g. attack or decay; Means for producing special musical effects, e.g. vibratos or glissandos
- G10H1/06—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour
- G10H1/12—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour by filtering complex waveforms
- G10H1/125—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour by filtering complex waveforms using a digital filter
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H1/00—Details of electrophonic musical instruments
- G10H1/0091—Means for obtaining special acoustic effects
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K15/00—Acoustics not otherwise provided for
- G10K15/08—Arrangements for producing a reverberation or echo sound
- G10K15/12—Arrangements for producing a reverberation or echo sound using electronic time-delay networks
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2210/00—Aspects or methods of musical processing having intrinsic musical character, i.e. involving musical theory or musical parameters or relying on musical knowledge, as applied in electrophonic musical tools or instruments
- G10H2210/155—Musical effects
- G10H2210/265—Acoustic effect simulation, i.e. volume, spatial, resonance or reverberation effects added to a musical sound, usually by appropriate filtering or delays
- G10H2210/281—Reverberation or echo
- G10H2210/291—Reverberator using both direct, i.e. dry, and indirect, i.e. wet, signals or waveforms, indirect signals having sustained one or more virtual reflections
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
- G10H2250/041—Delay lines applied to musical processing
- G10H2250/046—Delay lines applied to musical processing with intermediate taps
Definitions
- the present invention relates to the processing of digitised audio signals and particularly, although not exclusively, to the handling of processor resources during the addition of echo or reverberation effects by such processing.
- an audio signal processor involves a reverberation network for processing including a plurality of wave-ladder filters, each of which simulates a corresponding reflected audio signal from wave propagation in a selected direction.
- a reverberation network for processing including a plurality of wave-ladder filters, each of which simulates a corresponding reflected audio signal from wave propagation in a selected direction.
- Each of these filters receives as its input a signal derived from the original audio signal and the outputs of the filters are combined in an adder to create the reverberation effect.
- a reverberation imparting device comprising a frequency characteristics control circuit for controlling frequency characteristics of a reverberation sound and frequency characteristics setting means for timewise changing filter characteristics of the frequency characteristics control circuit.
- An example of the frequency characteristics control circuit is disclosed in which past samples of an input signal are read out within one sampling period and applied to two channels A and B and provided by multipliers 258 and 260 with the coefficients of filter characteristics. The respective output data of each multiplier is sequentially accumulated; the accumulated values are crossfaded so that the filter characteristics are gradually changed from those of the channel A to those of the channel B.
- the filtering may be applied by one or more suitably programmed processors which can vary the treatments applied in response to user or system commands. Filtering can, however, prove relatively computationally expensive and, for relatively low powered systems the processor will be working at or near peak capacity for the majority of time. Unless particular measures are provided to avoid it, changes in the setting of the filters can either lead to unacceptable delays as the processor runs out of capacity to handle the changing instructions in real time, leading to audible "clicks" appearing in the output signal, or a more powerful processor has to be used with the capability to handle these short-term requirements for additional processing capability.
- digital audio signal processing apparatus comprising a digital delay line connected to receive an input audio signal and having a plurality of outputs, each at a respective delay with respect to the input signal; a respective signal filter coupled to each delay line output, each filter having at least one variable setting; and a mixer stage arranged to receive and combine the outputs of the signal filters and to output a processed audio signal;
- the signal filters are provided by a programmable digital data processor arranged to receive the delay line outputs and process each according to a respective current filter setting and wherein, on receipt of instructions to change a filter setting for a particular delay line output, the processor generates a second filter for that output in parallel with the first filter and at the new setting, filters the output at both the old and new settings, and outputs a signal fading from the first filter the second filter output and wherein, on receipt of instructions to change a plurality of filter settings for respective delay line outputs, the processor creates only so many respective parallel filters as can be accommodated at one time within predetermined restrictions on
- the processor is preferably programmed to create only so many respective parallel filters as can be accommodated at one time within predetermined restrictions on the data handling capability of the processor.
- the processor may change the settings of the filters, including the creation of respective parallel filters and fading between outputs thereof, one after the other.
- the apparatus may further comprise two or more audio signal inputs each with a respective delay line, with the output filters for all delay lines being provided by the same processor, which processor is arranged to allow changes to filter settings for only one of the delay lines at a time - again preventing overloading of the processor. With not all filter settings being changed simultaneously, the apparatus may further contain storage means holding an instruction queue in which are placed those instructions to change one or more filter settings that are not immediately acted upon.
- the apparatus may further comprise user-operable input means coupled with the processor and by means of which a user can specify, directly or indirectly, changes to one or more filter settings.
- the mixer stage may be further coupled to receive the or each input signal and arranged to combine this with the outputs of the signal filters to output said processed audio signal and/or coupled to receive one or more further digital audio signals and arranged to combine the or each such signal with the outputs of the signal filters to output said processed audio signal.
- this effect is achieved by forming an output (OUT) signal which consists of a so-called DRY signal (the microphone output following such analogue to digital conversion and other localised signal processing as necessary) summed with an additional signal component formed from a weighted sum of filtered and delayed versions of the dry signal.
- OUT output
- DRY the microphone output following such analogue to digital conversion and other localised signal processing as necessary
- SA initial setting 18
- SB initial setting 14
- SC 14
- SD initial setting 18
- the outputs from the filter stages 12, 14, 16, 18 are summed by mixer 20 together with the dry signal.
- a memory 22 which acts as a queue Q or buffer for user or system commands UIP to change one or more of the filter settings SA, SB, SC, SD.
- FIG. 2 represents the functional components of an apparatus embodying the present invention in a twin-microphone karaoke apparatus.
- Each of the two microphones 24, 26 outputs its signal to a respective analogue to digital (A/D) converter 28, 30 and from there the respective digital bit streams are read into respective multiple-tapped random access memories MEM 32, 34.
- Each of these memories MEM is coupled with a programmed digital data processor PROC 36 and forms a respective digital delay line through reading out of the contents thereof at predetermined times D(1), D(2)...D(n) after they were read in.
- the processor 36 may be a slave unit operating under control of a central control and processing unit (CPU) 38 for the apparatus as a whole, or the functions of processor 36 may be carried out by CPU 38 as a dedicated subset of its functions.
- CPU central control and processing unit
- the processor stage 36 applies the treatments specified for each of the filter stages 12, 14, 16, 18 (Fig.1) to the digital data read from the memories 32, 34 and outputs this as a digital audio stream to mixer stage 40: as in the case of the mixer 20 in the Figure 1 embodiment, the output signal may include an unprocessed (or dry) input signal component routed directly through the memories 32 or 34 without delay and via the processor 36 to the mixer 40.
- a source of background music read from optical disc 42 which music the holders of microphones 24, 26 are singing along with:
- the visual display of lyrics for the singers on a display screen is a well-known feature of such systems and, since it bears no relation to operation in accordance with the requirements of the present invention, it is neither illustrated nor discussed further.
- the desired track of background digital audio that the user wishes to sing along with is indicated to the apparatus by means of user input controls 44, which controls may comprise a menu selection from a touch screen display, the input of an alphanumeric code to indicate the selection as one from a group or in other ways as will be readily understood.
- the digital audio is read from the disc 42 by disc reader 46 under direction of the system CPU 38.
- the audio on disc will generally be in encoded form (e.g. MPEG)
- the digital audio read by reader 46 is suitably passed via a decoder stage 48 before being read, in decoded form, into a buffer 50. From the buffer 50, any necessary signal treatments such as pitch shifting 52 are applied under direction of the CPU 38 before the background audio is added as a further component to the mixer 40.
- the output digital audio signal is subjected to any such further signal processing as required (at stage 54) before being output to amplifier 56 and from thence to left and right speakers 58, 60 respectively: note that amplifier 56 is presumed to be a digital signal amplifier and hence no separate digital to analogue (D/A) stage is shown.
- D/A digital to analogue
- Figure 4 shows the situation as the setting SD of filter F1 is changed to setting SE.
- the change involves the creation by the hosting processor of a second F1 filter 112 in parallel with the first 12, the second filter being at the new setting SE.
- the tapped signal from delay line 10 is passed to both copies of the filter and, as indicated by arrow 62, a cross fade is applied on the outputs to mixer 20 such as to give a smooth transition from the first setting SD to the second SE.
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- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Algebra (AREA)
- General Physics & Mathematics (AREA)
- Mathematical Analysis (AREA)
- Mathematical Optimization (AREA)
- Mathematical Physics (AREA)
- Pure & Applied Mathematics (AREA)
- Theoretical Computer Science (AREA)
- Signal Processing (AREA)
- Reverberation, Karaoke And Other Acoustics (AREA)
Abstract
Description
- The present invention relates to the processing of digitised audio signals and particularly, although not exclusively, to the handling of processor resources during the addition of echo or reverberation effects by such processing.
- One example of an audio signal processor is given in US Patent 5,774,560 and involves a reverberation network for processing including a plurality of wave-ladder filters, each of which simulates a corresponding reflected audio signal from wave propagation in a selected direction. Each of these filters receives as its input a signal derived from the original audio signal and the outputs of the filters are combined in an adder to create the reverberation effect.
- In US patent 4,706,291 there is disclosed a reverberation imparting device comprising a frequency characteristics control circuit for controlling frequency characteristics of a reverberation sound and frequency characteristics setting means for timewise changing filter characteristics of the frequency characteristics control circuit. An example of the frequency characteristics control circuit is disclosed in which past samples of an input signal are read out within one sampling period and applied to two channels A and B and provided by multipliers 258 and 260 with the coefficients of filter characteristics. The respective output data of each multiplier is sequentially accumulated; the accumulated values are crossfaded so that the filter characteristics are gradually changed from those of the channel A to those of the channel B.
- Further examples may be found in the field of karaoke systems where echo/reverberation effects are often applied to the signal from a singers microphone. The effect is achieved by forming an output signal which consists of a generally unmodified signal (a so-called "dry" signal) summed with an additional signal component formed from a weighted sum of filtered and delayed versions of the dry signal, with the filtering providing the reverberation and the delay producing the echo. Such a system will be described hereinafter with reference to Figure 1 of the attached drawings.
- As will be readily understood, for digitised data, the filtering may be applied by one or more suitably programmed processors which can vary the treatments applied in response to user or system commands. Filtering can, however, prove relatively computationally expensive and, for relatively low powered systems the processor will be working at or near peak capacity for the majority of time. Unless particular measures are provided to avoid it, changes in the setting of the filters can either lead to unacceptable delays as the processor runs out of capacity to handle the changing instructions in real time, leading to audible "clicks" appearing in the output signal, or a more powerful processor has to be used with the capability to handle these short-term requirements for additional processing capability.
- It is accordingly an object of the present invention to provide an audio signal processing apparatus having a plurality of programmable filters which avoids audio discontinuities as filter settings are changed.
- It is a further object to provide such an apparatus which may achieve this without requiring excessive processor capacity for the filters.
- In accordance with the present invention there is provided digital audio signal processing apparatus comprising a digital delay line connected to receive an input audio signal and having a plurality of outputs, each at a respective delay with respect to the input signal;
a respective signal filter coupled to each delay line output, each filter having at least one variable setting; and
a mixer stage arranged to receive and combine the outputs of the signal filters and to output a processed audio signal;
wherein the signal filters are provided by a programmable digital data processor arranged to receive the delay line outputs and process each according to a respective current filter setting and wherein, on receipt of instructions to change a filter setting for a particular delay line output, the processor generates a second filter for that output in parallel with the first filter and at the new setting, filters the output at both the old and new settings, and outputs a signal fading from the first filter the second filter output and wherein, on receipt of instructions to change a plurality of filter settings for respective delay line outputs, the processor creates only so many respective parallel filters as can be accommodated at one time within predetermined restrictions on the data handling capability of the processor. By the creation of a second filter, a smooth transition between the settings may be achieved. - In order that this creation of additional filters for the duration of a setting change does not create excessive processing overheads, on receipt of instructions to change a plurality of filter settings for respective delay line outputs, the processor is preferably programmed to create only so many respective parallel filters as can be accommodated at one time within predetermined restrictions on the data handling capability of the processor. In a particularly processor-restricted embodiment, the processor may change the settings of the filters, including the creation of respective parallel filters and fading between outputs thereof, one after the other.
- The apparatus may further comprise two or more audio signal inputs each with a respective delay line, with the output filters for all delay lines being provided by the same processor, which processor is arranged to allow changes to filter settings for only one of the delay lines at a time - again preventing overloading of the processor. With not all filter settings being changed simultaneously, the apparatus may further contain storage means holding an instruction queue in which are placed those instructions to change one or more filter settings that are not immediately acted upon.
- The apparatus may further comprise user-operable input means coupled with the processor and by means of which a user can specify, directly or indirectly, changes to one or more filter settings. Additionally, the mixer stage may be further coupled to receive the or each input signal and arranged to combine this with the outputs of the signal filters to output said processed audio signal and/or coupled to receive one or more further digital audio signals and arranged to combine the or each such signal with the outputs of the signal filters to output said processed audio signal.
- Further features and advantages of the present invention will become apparent from reading of the following description of preferred embodiments of the invention, given by way of example only, and with reference to the accompanying drawings in which:
- Figure 1 schematically represents a multiple output digital delay line with output filters on each output for the creation of reverberation and echo effects;
- Figure 2 is a block schematic diagram of a data processing apparatus suitable to embody the present invention;
- Figures 3 to 6 sequentially represent the changing of filter settings in an apparatus embodying the present invention; and
- Figure 7 illustrates the sequential handling of filter change operations with multiple delay lines.
- In the following description, a dual microphone karaoke system embodying the present invention will be described with reference to the attached Figures 1 to 7: it will be understood, however, that the present invention is not limited to such applications and may be used in other digital audio signal processing systems such as echo cancellation circuits in telecommunications applications, as will be readily understood by the skilled reader.
- As previously mentioned, in karaoke systems echo and/or reverberation effects are often applied to the microphone input. As shown in Figure 1, this effect is achieved by forming an output (OUT) signal which consists of a so-called DRY signal (the microphone output following such analogue to digital conversion and other localised signal processing as necessary) summed with an additional signal component formed from a weighted sum of filtered and delayed versions of the dry signal. The echo results from passing the dry signal through a multiple-tapped
delay line 10 where the amount of available delay ranges from T=0 (no delay) to T=End (maximum delay) with tapping outputs being shown for four increasing delay values, D(1), D(2), D(n-1), and D(n): other values between D(2) and D(n-1) may of course be inserted. On each of these outputs, providing reverberation effects, are respective filter stages 12 (F1), 14 (F2), 16 (Fn-1), 18 (Fn), with each filter having a respective initial setting 18 (SA), 16 (SB), 14 (SC) and 12 (SD). In order to produce the output signal OUT, the outputs from thefilter stages mixer 20 together with the dry signal. - Also shown coupled to each of the
filter stages memory 22 which acts as a queue Q or buffer for user or system commands UIP to change one or more of the filter settings SA, SB, SC, SD. - Figure 2 represents the functional components of an apparatus embodying the present invention in a twin-microphone karaoke apparatus. Each of the two
microphones converter access memories MEM data processor PROC 36 and forms a respective digital delay line through reading out of the contents thereof at predetermined times D(1), D(2)...D(n) after they were read in. As shown, theprocessor 36 may be a slave unit operating under control of a central control and processing unit (CPU) 38 for the apparatus as a whole, or the functions ofprocessor 36 may be carried out byCPU 38 as a dedicated subset of its functions. - The
processor stage 36 applies the treatments specified for each of thefilter stages memories mixer 20 in the Figure 1 embodiment, the output signal may include an unprocessed (or dry) input signal component routed directly through thememories processor 36 to themixer 40. - Also connected as input to the
mixer 40 for the illustrated karaoke embodiment is a source of background music read fromoptical disc 42, which music the holders ofmicrophones - The desired track of background digital audio that the user wishes to sing along with is indicated to the apparatus by means of
user input controls 44, which controls may comprise a menu selection from a touch screen display, the input of an alphanumeric code to indicate the selection as one from a group or in other ways as will be readily understood. Following selection, the digital audio is read from thedisc 42 bydisc reader 46 under direction of thesystem CPU 38. As the audio on disc will generally be in encoded form (e.g. MPEG), the digital audio read byreader 46 is suitably passed via adecoder stage 48 before being read, in decoded form, into abuffer 50. From thebuffer 50, any necessary signal treatments such as pitch shifting 52 are applied under direction of theCPU 38 before the background audio is added as a further component to themixer 40. - From the
mixer 40, the output digital audio signal is subjected to any such further signal processing as required (at stage 54) before being output toamplifier 56 and from thence to left andright speakers amplifier 56 is presumed to be a digital signal amplifier and hence no separate digital to analogue (D/A) stage is shown. - The arrangement whereby the settings of one or more of the filters (12, 14, 16, 18; Fig.1) changes setting will now be described with reference to the sequence of drawings in Figures 3 to 6. In Figure 3, a command to change settings UIP has been received and queued in
buffer 22. The change command is communicated to the filters affected (F1, F2 in this example) and then either due to restrictions in the available processor capability in the processor (38; Fig.2) hosting the filters, or as a matter of routine practise for the system, it is decided to change the settings sequentially. - Figure 4 shows the situation as the setting SD of filter F1 is changed to setting SE. The change involves the creation by the hosting processor of a second F1
filter 112 in parallel with the first 12, the second filter being at the new setting SE. The tapped signal fromdelay line 10 is passed to both copies of the filter and, as indicated byarrow 62, a cross fade is applied on the outputs to mixer 20 such as to give a smooth transition from the first setting SD to the second SE. - In Figure 5, the change from setting SD to SE for the F1 filter has been concluded and the
copy 12 of the filter at the SD setting has been deleted. The next (F2)filter 14 is being changed from setting SC to SF and again aparallel filter 114 is created at the new setting. Also as before, there is a cross-fade (indicated by arrow 64) to give a smooth transition between the settings. - In Figure 6, the settings change has been concluded and only those copies of the filters F1, F2 at the new settings (SE, SF respectively) remain. As will be clear from the progress of Figures 3 to 6, overloads in required processing capability are avoided by scheduling the filter changes to occur sequentially. Where there are a larger number of filters to change settings for, and sufficient processing capability to manage without slowing operations unacceptably, the filters may be changed in small groups (perhaps two or three filters) simultaneously, with the groups being handled sequentially.
- In general, changing the individual filter settings sequentially is preferred to give an even loading to the processor. This may also apply in situations as shown in Figure 7 where there are two
delay lines 10, 110 (a separate one for each microphone MIC1, MIC2) and it is desired to change the settings for thefilters delay line 10 and thefilters second delay line 110. Where the filters are all provided by the same processor, the rule that only one may change at a time is applied, with the filters being treated in theorder delay line 10 output being changed before the settings for thedelay line 110 output. - From reading the present disclosure, other variations (including conformance or otherwise with other optical or magnetic disc recording standards) will be apparent to persons skilled in the art. Such variations may involve other features which are already known in the field of apparatuses for replaying of audio and/or video signals and component parts thereof and which may be used instead of or in addition to features already described herein.
Claims (7)
- Digital audio signal processing apparatus comprising a digital delay line (10) connected to receive an input audio signal (DRY) and having a plurality of outputs, each at a respective delay with respect to the input signal;
a respective signal filter (12, 14, 16, 18) coupled to each delay line output, each filter having at least one variable setting; and
a mixer stage 20 arranged to receive and combine the outputs of the signal filters (12, 14, 16, 18) and to output a processed audio signal;
wherein the signal filters (12, 14, 16, 18) are provided by a programmable digital data processor (36, 38) arranged to receive the delay line outputs and process each according to a respective current filter setting and wherein, on receipt of instructions to change a filter setting for a particular delay line output, the processor generates a second filter (112) for that output in parallel with the first filter (12) and at the new setting, filters the output at both the old and new settings, and outputs a signal fading from the first filter (12) to the second filter (112) output characterized in being arranged such that on receipt of instructions to change a plurality of filter settings for respective delay line outputs, the processor (36, 38) creates only so many respective parallel filters as can be accommodated at one time within predetermined restrictions on the data handling capability of the processor (36, 38). - Apparatus as claimed in Claim 1, wherein on receipt of instructions to change a plurality of filter settings for respective delay line outputs, the processor (36, 38) changes the settings of the filters, including the creation of respective parallel filters and fading between outputs thereof, one after the other.
- Apparatus as claimed in Claim 1 or Claim 2, further comprising two or more audio signal inputs (MIC1, MIC2) each with a respective delay line (10, 110), wherein the output filters for all delay lines are provided by the same processor (36, 38), which processor (36, 38) is arranged to allow changes to filter settings for only one of the delay lines at a time.
- Apparatus as claimed in Claim 3, further comprising storage means (22) holding an instruction queue in which are placed those instructions to change one or more filter settings that are not immediately acted upon.
- Apparatus as claimed in any of Claims 1 to 4, further comprising user-operable input means (44) coupled with the processor and by means of which a user can specify, directly or indirectly, changes to one or more filter settings.
- Apparatus as claimed in any of Claims 1 to 5, wherein the mixer stager (20) further coupted to receive the or each input signal (DRY) and arranged to combine this with the outputs of the signal filters (12, 14, 16, 18) to output said processed audio signal.
- Apparatus as claimed in any of Claims 1 to 6, wherein the mixer stage (20) further coupled to receive one or more further digital audio signals and arranged to combine the or each such signal with the outputs of the signal filters (12, 14, 16, 18) to output said processed audio signal.
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
GBGB9920326.7A GB9920326D0 (en) | 1999-08-28 | 1999-08-28 | Digital audio signal processing |
GB9920326 | 1999-08-28 | ||
PCT/EP2000/007923 WO2001016934A1 (en) | 1999-08-28 | 2000-08-14 | Digital audio signal processing apparatus comprising a delay line |
Publications (2)
Publication Number | Publication Date |
---|---|
EP1125274A1 EP1125274A1 (en) | 2001-08-22 |
EP1125274B1 true EP1125274B1 (en) | 2006-03-15 |
Family
ID=10859929
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP00960453A Expired - Lifetime EP1125274B1 (en) | 1999-08-28 | 2000-08-14 | Digital audio signal processing apparatus comprising a delay line |
Country Status (6)
Country | Link |
---|---|
EP (1) | EP1125274B1 (en) |
JP (1) | JP2003508803A (en) |
KR (1) | KR100717519B1 (en) |
DE (1) | DE60026719T2 (en) |
GB (1) | GB9920326D0 (en) |
WO (1) | WO2001016934A1 (en) |
Families Citing this family (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP2418773A1 (en) * | 2010-07-28 | 2012-02-15 | Brandenburgische Technische Universität Cottbus | Glitch-free switchable FIR-filter |
Family Cites Families (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4706291A (en) * | 1985-06-25 | 1987-11-10 | Nippon Gakki Seizo Kabushiki Kaisha | Reverberation imparting device |
JP3296648B2 (en) * | 1993-11-30 | 2002-07-02 | 三洋電機株式会社 | Method and apparatus for improving discontinuity in digital pitch conversion |
JP2746157B2 (en) * | 1994-11-16 | 1998-04-28 | ヤマハ株式会社 | Electronic musical instrument |
US5774560A (en) * | 1996-05-30 | 1998-06-30 | Industrial Technology Research Institute | Digital acoustic reverberation filter network |
-
1999
- 1999-08-28 GB GBGB9920326.7A patent/GB9920326D0/en not_active Ceased
-
2000
- 2000-08-14 WO PCT/EP2000/007923 patent/WO2001016934A1/en active IP Right Grant
- 2000-08-14 KR KR1020017005302A patent/KR100717519B1/en not_active IP Right Cessation
- 2000-08-14 JP JP2001520400A patent/JP2003508803A/en active Pending
- 2000-08-14 DE DE60026719T patent/DE60026719T2/en not_active Expired - Fee Related
- 2000-08-14 EP EP00960453A patent/EP1125274B1/en not_active Expired - Lifetime
Also Published As
Publication number | Publication date |
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DE60026719T2 (en) | 2006-10-12 |
KR100717519B1 (en) | 2007-05-14 |
JP2003508803A (en) | 2003-03-04 |
GB9920326D0 (en) | 1999-11-03 |
KR20010089369A (en) | 2001-10-06 |
WO2001016934A1 (en) | 2001-03-08 |
EP1125274A1 (en) | 2001-08-22 |
DE60026719D1 (en) | 2006-05-11 |
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