EP0101895A2 - Dispositif et procédé pour générer la réverbération des signaux tonales analogues - Google Patents

Dispositif et procédé pour générer la réverbération des signaux tonales analogues Download PDF

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Publication number
EP0101895A2
EP0101895A2 EP83107156A EP83107156A EP0101895A2 EP 0101895 A2 EP0101895 A2 EP 0101895A2 EP 83107156 A EP83107156 A EP 83107156A EP 83107156 A EP83107156 A EP 83107156A EP 0101895 A2 EP0101895 A2 EP 0101895A2
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EP
European Patent Office
Prior art keywords
audio data
digital
microprocessor
audio signals
multiplications
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.)
Granted
Application number
EP83107156A
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German (de)
English (en)
Other versions
EP0101895B1 (fr
EP0101895A3 (en
Inventor
Reinhard Franz
Wilfried Dipl.-Ing. Dittmar
Christian Dipl.-Phys. Scheidegger
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
FRANZ, REINHARD
Original Assignee
Wersi Electronic GmbH and Co KG
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Publication date
Application filed by Wersi Electronic GmbH and Co KG filed Critical Wersi Electronic GmbH and Co KG
Priority to AT83107156T priority Critical patent/ATE41714T1/de
Publication of EP0101895A2 publication Critical patent/EP0101895A2/fr
Publication of EP0101895A3 publication Critical patent/EP0101895A3/de
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Publication of EP0101895B1 publication Critical patent/EP0101895B1/fr
Expired legal-status Critical Current

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    • 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

Definitions

  • the invention relates to a device for generating Hall for analogue audio signals, with an A / D converter for converting the analogue audio signals into digital audio data, a processing unit for handling the digital audio data, a D / A converter for converting back the treated digital audio data and a microprocessor, wherein the processing unit comprises a T onquel memory and a computer for additions and multiplications, which are connected to lines for the digital audio data, and a method for operating this device.
  • the desired reverberation is generated by performing several all-pass filter calculations one after the other, with the sound data that have been stored at different times being added after they have been partially before have been multiplied by a gain factor of less than 1. All arithmetic operations must be carried out within one working cycle of the converter. Since the clock times have to be kept short for the flawless replication of the analog signals by digital data, but on the other hand the computing processes require a large number of individual steps due to the numerous multiplications, the processing unit has a computer designed as an arithmetic processor with a very high clock frequency and sound data adapted to this frequency Memory. Both are very complex assemblies, so that the Hall device is very expensive overall.
  • a microprocessor which is connected to a remote control part, the time control of the processing unit, a program memory part and possibly further control devices. It is used to select and influence the reverb program, taking into account the settings of the remote control part.
  • the invention has for its object to provide a device of the type described above, which has a much simpler structure for generating the desired Hall and can therefore be manufactured cheaper.
  • microprocessor is connected to lines for the digital audio data and is used as a computer which carries out the multiplications by shifting and, if appropriate, adding digital values.
  • Microprocessors have a comparative one low clock frequency. If they are to be used to multiply two digital values, the transducers could be carried out at most one or two multiplications within one work cycle, which is insufficient for the Hall generation. However, if the multiplication is limited to certain values, namely those which result from shifting and possibly adding, that is to say summing or subtracting, these multiplications require only a very few steps.
  • the clock frequency of a conventional microprocessor is therefore sufficient to carry out the required number of multiplications and other arithmetic operations within one working cycle of the converter.
  • the microprocessor not only performs control tasks, but also works as a computer, the previously used special computer (arithmetic processor) with a high clock frequency can be omitted.
  • microprocessor is designed for internal 16 bit processing. Since it is necessary to use at least 8 bit audio data to adequately resolve the analog signals, the shifting and adding of the digital values can be carried out in one step.
  • the converters are designed for processing 12 to 14 bit audio data. This results in a particularly accurate replication of the analog audio signals using digital audio data.
  • the microprocessor with the internal 16 bit processing can also carry out the necessary calculations with this 12 to 14 bit audio data.
  • the microprocessor has connections for 8 audio data lines. 12 bit audio data is then transmitted in two steps using a single instruction.
  • the microprocessor for calling up various Hall programs from a program memory is provided with an indexing device which has a reset connection (Reset) which is used to actuate the first program and an indexing connection (Interrupt), which is used to control the next program.
  • Reset reset connection
  • Interrupt indexing connection
  • a squelch circuit for suppressing the smallest amplitudes can be provided in the output circuit of the analog audio signals. This suppresses the output signals as soon as a predetermined level is undershot. This prevents unwanted quantization noise and disturbing quantization distortion.
  • a method for operating this device in which several all-pass filter calculations are carried out by multiplication with an amplification factor of less than 1, is characterized in that all multiplications are carried out with the same amplification factor. In this way, all digital audio data can be treated in the same way, which leads to considerable simplifications.
  • the multiplications should be made with the gain factor 0.75.
  • the gain factor 0.75.
  • this value leads to very appealing Hall results.
  • gain factors such as 0.5 or 0.625 or 0.875 u.
  • a further advantageous measure consists in that the digital sound data occurring successively on the A / D converter is read into the sound data memory under cyclically recurring addresses, the addresses of the sound data to be called up for the individual all-pass filter calculations by means of a switchable base number or auxiliary numbers differing from this are determined by permanently entered subtraction values, the basic number is cyclically advanced to double the address number of the sound signal memory and the basic or auxiliary numbers shortened by the first binary position are used as addresses.
  • an analog audio signal is fed via an input 1 to an emphasis circuit EM which produces a treble boost and a bass attenuation in order to achieve the A / D at all frequencies of this analog input signal at the maximum amplitudes to be expected in each case. and exploit D / A converter accuracy.
  • the analog sound signal is then fed to a compressor circuit Co, in which the sound signal is compressed by 10 dB between -30 dB and -50 dB for the purpose of physiological adaptation.
  • a filter circuit F1 follows, which has a six-pole Chebyshev low-pass filter to meet the condition of the sampling theorem (band limitation).
  • a sample and hold circuit SH and an analog-to-digital converter AD follow.
  • the sample and hold circuit SH ensures that a stable signal is present at the A / D converter input during the A / D conversion.
  • the conversion takes place with a 12 bit module at a clock frequency of 10 kHz, for example.
  • the processing unit 2 includes a microprocessor MP, to which a program memory PM is assigned, and a sound data memory SM.
  • a 2k x 8 ROM can be used as the program memory
  • an 8k x 8 RAM can be used as the sound data memory.
  • a digital-to-analog converter DA is provided on the output side, which in turn is a 12-bit module.
  • a de-emphasis circuit DE follows, which compensates for the effect of the emphasis circuit EM, so that the overall frequency response becomes linear.
  • a filter circuit F2 which has a six-pole Chebyshev low-pass filter at 4 kHz due to the sampling theorem.
  • a squelch circuit SC is provided, to which the output line 3 for the analog audio signal connects.
  • the microprocessor MP has eight data connections D, which are connected to an 8 bit data bus 4. This data bus is connected to the analog-digital converter AD, the digital-analog converter DA, the sound data memory SM and the program memory PM and therefore not only carries program signals but also serves as a sound data line before, during and after processing. There are also twelve address connections A, which is connected to the program memory PM and the sound signal memory SM via a 12 bit address bus 5.
  • the control connections C are connected via a control bus 6 to the sample and hold circuit SH, the analog-to-digital converter AD, the digital-to-analog converter DA, the program memory PM and the sound data memory SM. For reasons of simplification, decoders or interfaces provided are also not shown.
  • the microprocessor MP has an internal 16-bit processing and can be operated, for example, with an internal computing speed of 2 MHz. He also owns a step-up device 7 in order to let various Hall programs from the program memory PM take effect. For this purpose, a reset connection R and a relay connection I are provided. If a pulse is applied to the reset connection R, the first program is activated from a program cycle. Each time a pulse is applied to the switching terminal I, the next program in the cycle is switched to.
  • FIG. 2 and 3 explain how the digital sound data obtained in the analog-digital converter AD are treated in the processing unit 2.
  • Fig. 2 shows the calculation process for an all-pass filter.
  • Fig. 3 shows that four such filter calculations are carried out in succession.
  • the delayed signal s2 is generated by multiplying V 1 of a signal s4 by the factor k.
  • the signal s4 is taken from the sound data memory SM at an address z -n . n indicates by how many converter work cycles the sound signal s3 was previously written into the memory SM.
  • the delayed signal s4 is subjected to an addition A2 with a signal s5, which was generated by multiplying V2 of the signal s3 by the factor -k.
  • the emerging sum signal s6 is then processed further.
  • the gain factor k always has the same value 0.75. This is obtained, for example, by shifting the respective sound data by one binary digit and again by one binary digit (which corresponds to multiplication by a factor of 0.5) and then summing the two shifted values.
  • the reverberation time can be set between 0.1 and 5 s, so that all sensible reverberation times that are commonly used in music today can be achieved with this system.
  • FIG. 4 schematically shows in the area I those addresses under which sound data can be stored in the memory SM. For example, these addresses go from 00000 to 11111. In area II, this memory is shown once again with the same addresses, which has the same addresses. A base number B, which has another position in front of the address, is advanced in the direction of the arrow x. By subtracting with permanently entered values, several auxiliary numbers H 1 , H 2 and H 3 are linked to the base number B, which therefore experience the same increment as the base number. The base number B is counted up to twice the memory content, ie to the end of area II.
  • the exemplary embodiment works with values such as can be achieved with commercially available microprocessors. As much as newer microprocessors have higher clock frequencies, multiplications can also be carried out in which the shifting and adding requires more steps or more than four all-pass filter calculations are carried out in succession. The multiplications also include measures in which shifted signals are added not by summation but by subtraction, for example from the original signal.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Analogue/Digital Conversion (AREA)
  • Measurement Of Velocity Or Position Using Acoustic Or Ultrasonic Waves (AREA)
  • Electrophonic Musical Instruments (AREA)
  • Reverberation, Karaoke And Other Acoustics (AREA)
EP83107156A 1982-08-27 1983-07-21 Dispositif et procédé pour générer la réverbération des signaux tonales analogues Expired EP0101895B1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AT83107156T ATE41714T1 (de) 1982-08-27 1983-07-21 Vorrichtung zur erzeugung von hall fuer analoge tonsignale und verfahren zu deren betrieb.

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE3231925A DE3231925C2 (de) 1982-08-27 1982-08-27 Vorrichtung zur Erzeugung von Hall für analoge Tonsignale und Verfahren zu deren Betrieb
DE3231925 1982-08-27

Publications (3)

Publication Number Publication Date
EP0101895A2 true EP0101895A2 (fr) 1984-03-07
EP0101895A3 EP0101895A3 (en) 1985-06-19
EP0101895B1 EP0101895B1 (fr) 1989-03-22

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EP83107156A Expired EP0101895B1 (fr) 1982-08-27 1983-07-21 Dispositif et procédé pour générer la réverbération des signaux tonales analogues

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EP (1) EP0101895B1 (fr)
AT (1) ATE41714T1 (fr)
DE (1) DE3231925C2 (fr)

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE19733300C2 (de) * 1997-08-01 2001-04-19 Joachim Hecht Verfahren zur elektronischen Nachbildung der Übertragungseigenschaften eines ein Band-Echohall-Gerät umfassendes Klang-Effekt-Gerätes mittels eines programmgesteuerten digitalen oder analogen Signalprozessors

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2277488A1 (fr) * 1974-07-03 1976-01-30 Polygram Gmbh Procede et dispositif pour realiser une reverberation artificielle

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2277488A1 (fr) * 1974-07-03 1976-01-30 Polygram Gmbh Procede et dispositif pour realiser une reverberation artificielle

Non-Patent Citations (3)

* Cited by examiner, † Cited by third party
Title
POLYTECHNISCH TIJDSCHRIFT GELUID EN TRILLING, Band 35, Juni 1980, Seiten 80-84, DELFT, (NL). D. GRIESINGER:"Programmeerbare digitale nagalm". * SEITE 81, SPALTE 1, ZWEILE 18 - SPALTE 3, ZEILE 28; FIGUR 1 * *
RADIOMENTOR ELECTRONIC, Band 42, Nr.1, Januar 1976, Seiten 023-026, BERLIN, (DE). K. BADER et al.:"Klangumformung durch Computer". * SEITE 024, SPALTE 1, ZEILEN 3-30; SPALTE 3, ZEILE 33 - SEITE 025, SPALTE 3, ZEILE 32; FIGUREN 1,3 * *
STUDIO (Sonderdruck Sept. 80), AUDIO Export Georg NEUMANN & Co., GmbH (LEXIKON 224 DIGITALES HALLGERÄT) *

Also Published As

Publication number Publication date
DE3231925A1 (de) 1984-03-01
EP0101895B1 (fr) 1989-03-22
DE3231925C2 (de) 1986-04-17
EP0101895A3 (en) 1985-06-19
ATE41714T1 (de) 1989-04-15

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