US9271096B2 - Delay unit for a conference audio system, method for delaying audio input signals, computer program and conference audio system - Google Patents

Delay unit for a conference audio system, method for delaying audio input signals, computer program and conference audio system Download PDF

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US9271096B2
US9271096B2 US13/395,147 US200913395147A US9271096B2 US 9271096 B2 US9271096 B2 US 9271096B2 US 200913395147 A US200913395147 A US 200913395147A US 9271096 B2 US9271096 B2 US 9271096B2
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delay
audio
delegate
write
time delay
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US20120170768A1 (en
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Marc Smaak
C. P. Janse
Chen Tchang
L. C. A. van Stuivenberg
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Robert Bosch GmbH
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Robert Bosch GmbH
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M11/00—Telephonic communication systems specially adapted for combination with other electrical systems
    • H04M11/06—Simultaneous speech and data transmission, e.g. telegraphic transmission over the same conductors
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00—Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/009—Signal processing in [PA] systems to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/05—Application of the precedence or Haas effect, i.e. the effect of first wavefront, in order to improve sound-source localisation

Definitions

  • the invention relates to a delay unit allocated to a single delegate unit of a conference audio system adapted to delay audio input signals for an adjustable time delay, thereby generating audio output signals.
  • the invention relates to a delay unit allocated to a single delegate unit of a conference audio system adapted to delay audio input signals for an adjustable time delay, thereby generating audio output signals
  • the delay unit comprising a circular buffer, a write pointer to write a sample of a first audio input signal to the circular buffer at a first write position, a read pointer to read a sample from the circular buffer at a first read position as a sample of a first output signal, whereby the distance between the first write position and the first read position determines a first time delay, and a buffer control module adapted to move the write pointer to the next position after writing and to move the read pointer to a next position after reading.
  • the invention also relates to a method for delaying audio input signals, a respective computer program and a conference audio system comprising the delay unit.
  • conference audio systems also called sound reinforcement systems—comprise a central control unit and a plurality of so-called delegate units, which represent the working place of the delegates and have a microphone for inputting audio signals in the conference audio system and a loudspeaker for outputting audio signals.
  • delegate units which represent the working place of the delegates and have a microphone for inputting audio signals in the conference audio system and a loudspeaker for outputting audio signals.
  • Such a conference audio system is for example disclosed in EP 1 686 835, A1.
  • a speaker uses one of the microphones and the microphone signal is sent to the central control unit.
  • the signal is processed by the central control unit (like feedback suppression) and then distributed as a plurality of signals to the other delegate units.
  • the signals are passed to the loudspeakers as audio output signals, except for the delegate units where there is an active speaker in front of the delegate unit.
  • the loudspeaker signal is attenuated or is blocked to avoid howling.
  • a delay unit with the features of claim 1 a method for delaying input signals with the features of claim 7 , a computer program with the features of claim 8 and a conference audio system with the features of claim 9 are proposed.
  • Preferred or advantageous embodiments of the invention are disclosed by the dependent claims, the description and the figures as attached.
  • the length of the time delay is preferably chosen in accordance with the “Haas effect”.
  • the Haas effect is also called the precedence effect and describes the human psychoacoustic phenomena of correctly identifying the direction of the sound source heard in both ears. Due to the head's geometry (two ears spaced apart, separated by a barrier) the direct sound from any source first enters the ear closest to the source, then the ear farthest away.
  • the Haas effect describes that humans localize a sound source based upon the first arriving sound, if the subsequent arise within 25-35, ms delay. If the later arrivals are longer than this time delay, then two distinct sounds are heard.
  • a delay unit allocated to a single delegate unit of a conference audio system which is adapted to delay audio input signals for an adjustable time delay, thereby generating audio output signals.
  • the delay unit or units in the conference audio system is/are allocated to the individual delegate units, so that each delegate unit is provided with audio output signals with an individual time delay. Especially it is possible, that each individual time delay differs from the other.
  • the audio input signals are preferably provided by a microphone or another sound source, the audio output signals are intended to be output by the loudspeakers.
  • the delay unit comprises at least a circular buffer, which is defined by a logical memory architecture, whereby specific storage locations are used in an endless, ring-like manner.
  • a write pointer is provided to write a sample, for example a time piece, of a first input signal to the circular buffer at a first write position
  • a read pointer is provided to read a sample from the circular buffer at a first read position as a sample of a first output signal.
  • the distance between the first write position and the first read position which is the number of storage locations, each storage location being able to store a sample of the audio input signal, determines or represents a first time delay.
  • the time delay can be calculated by multiplying the number of storage locations with the temporal length of the sample.
  • a buffer control module is integrated, which is operable to move the write pointer to a next, especially following position after writing and to move the read pointer to a next, especially following position after reading.
  • a FIFO—first in first out—architecture is provided by the circular buffer and the buffer control module.
  • the buffer control module is adapted to adjust or set a second time delay by moving the write pointer to a second write position, whereby the distance between the first read position and the second write position determines the second time delay.
  • the second write position is in the said circular buffer.
  • a second circular buffer is provided and the write pointer is set to the second write position in the second circular buffer.
  • the read pointer may be set to the first read position in the second circular buffer.
  • the buffer control module is operable to change the time delay during or in connection with a change of the audio input signal source. Keeping in mind, that the length of the time delay is dependent on the distance and/or orientation of a sound source and the respective delegate unit, it is normally necessary to change the time delay as soon as the sound source and thus the distance and orientation changes.
  • the second write position is equal to the first write position and the write pointer does not change its place.
  • the remaining samples of the first audio input signal will be rendered followed by the samples of the second audio input signal.
  • the write pointer is readjusted, i.e. is put back and the samples of the second audio input signal are added to the samples of the first audio input signal at same memory locations. As a result all samples of the first audio input signal are output to the loudspeaker, partly overlapped by the samples of the second audio input signal.
  • the write pointer is readjusted, i.e. it is put forward, so that a number of memory locations are not filled by samples of the first audio input signal and the second audio input signal.
  • each memory location which is read out by the read pointer is afterwards set to zero, so that in this situation the samples of the first audio input signal will be rendered, followed by a number of zeros and then by the samples of the second audio input signal.
  • d old is smaller than B then only the last d old , samples can and should be sampled.
  • other fade-in and/or fade-out algorithms are possible.
  • the delay unit comprises a control module, which is adapted to store a lookup table or a map of possible audio sources and respective time delays or an equivalent data, so that the delay unit is capable to find the individual time delay for a specific audio source.
  • a further subject-matter of the invention relates to a method for delaying output input signals, which is preferably carried out by the delay unit as already described or according to one of the preceding claims.
  • a sample of a first audio input signal is written by a write pointer to a circular buffer at a first write position and a sample of the first audio input signal is read by a read pointer from the circular buffer at a first read position, whereby the distance between the first write position and the first read position determines the first time delay.
  • the write and read operations are performed in an endless manner, so that after writing the sample the write pointer is moved to a next position determined by (old position+1) mod N, whereby N is the length of the circular buffer.
  • the read pointer is moved to the next position which is determined by (old position+1) mod N after reading.
  • the write pointer is set to a second write position, whereby the distance between the first read position and the second write position determines the second time delay.
  • a further subject-matter of the invention relates to a computer program with the features of claim 8 .
  • a next subject-matter of the invention is a conference audio system—also called a sound reinforcement system with the features of claim 9 .
  • the conference audio system adds directivity to the output audio signals the system may also be called direction faithful sound (DFS) reinforcement system.
  • the conference audio system comprises a plurality of delegate units, each delegate unit having a delegate loudspeaker and/or a delegate microphone. As a fact it is possible that a delegate unit has both or has only a loudspeaker or a microphone.
  • a control means is employed for distributing at least one audio input signal from at least one of the delegate microphones or another sound source to a plurality of the delegate loudspeakers, whereby the plurality of delegate loudspeakers generate a common audio atmosphere.
  • delay means are provided, which are operable to add a time delay on the audio input signal.
  • the delay means are the delay unit or a plurality of such delay units according to one of the preceding claims 1 to 6 or as previously described.
  • the delay unit is positioned in the delegate units, in other embodiments it is also possible to centralize all or a part of the delay units for example in the control means and send the delayed audio input signals as audio output signals to the loudspeakers.
  • the time delay is dependent on the distance and/or direction between the position of the delegate microphone or sound source, respectively, generating the audio input signal and the individual delegate loudspeaker position.
  • each delay unit or at least the plurality of the delegate units have an individual time delay, which is different to the time delay of the adjacent and/or nearby delegate units.
  • FIG. 1 a schematic view of a congress audio system as a first embodiment of the invention
  • FIG. 2 a block diagram of a first possible realization of the congress audio system in FIG. 1 ;
  • FIG. 3 is a schematic diagram of a circular buffer employed in embodiments of the invention.
  • FIG. 4 is a schematic diagram of the circular buffer of FIG. 3 illustrating a case where a first audio delay is the same as a second audio delay.
  • FIG. 5 is a schematic diagram of the circular buffer of FIG. 3 illustrating a case where the second audio delay is longer than the first audio delay.
  • FIG. 6 is a schematic diagram of the circular buffer of FIG. 3 illustrating a case where the second audio delay is shorter than the first audio delay.
  • FIG. 1 shows a schematic view of a congress audio system 1 allowing a directional sound function based upon distributed loudspeakers.
  • the congress audio system 1 comprises a plurality of delegate units 2 , which are interconnected by a control means embodied as connection means 3 .
  • Most of the delegate units 2 comprise a delegate microphone 4 and a delegate loudspeaker 5 .
  • Some of the delegate units may only be realized as listener units 6 having only a delegate loudspeaker 5 or as speaker units 7 having only a delegate microphone 4 .
  • the delegate units 2 are integrated in a one-person workplace, for example realized as a lectern, a desktop or a seat for example in a congress hall, auditorium, lecture hall, courtroom or the like.
  • the delegate units 2 are for example arranged in rows and columns or in concentric circles.
  • the audio signal generated by an active delegate microphone 8 of a specific delegate unit 12 is provided with a time delay in dependence on the distance between the specific delegate unit 12 and the delegate unit 2 with the delegate loudspeaker 5 emitting the audio signal to the listeners.
  • the time delay is in accordance with the acoustic velocity (sound-propagation velocity).
  • the sound atmosphere of the listener imitates a directional sound resulting from the specific delegate unit 12 .
  • the human psychoacoustic phenomena of correctly identifying the direction of a sound source heard by both ears but arriving at different times is based on the Haas effect, also called the precedence effect.
  • a first time delay dl is added to the audio signal to be emitted by the delegate loudspeaker 5 of the delegate unit 9
  • a second time delay d 2 which is longer than the first delay d 1
  • a third time delay d 3 is added to the audio signal emitted by the delegate loudspeaker 5 of the delegate unit 11 , which is longer than the time delay d 2 and the time delay d 1 .
  • the listener of the delegate unit 10 also hears the emitted audio signals of the adjacent delegate units 9 and 11 and maybe further delegate units (not shown) he can identify a direction of a virtual sound source, whereby the direction of the virtual sound source is identical to the direction to the active microphone 8 .
  • the audio atmosphere of the listener at the delegate unit 10 is generated under participation of the delegate loudspeakers 5 of the delegate units 9 , 10 etc. next to the delegate unit 10 .
  • the sound from the adjacent delegate units 9 and 11 is significantly lower than the sound emitting from the delegate unit 10 it is still possible to recognize the direction of the virtual sound source, respectively the active microphone 8 , as the Haas effect is also true even in case the volume of the audio signals arriving at both ears of the listeners is different.
  • FIG. 2 shows a first possible embodiment of the congress audio system 1 comprising a plurality of the delegate units 2 .
  • connection means 3 is realized as a plurality of parallel channels, for example wires, whereby each delegate microphone 4 is connected to an individual microphone channel 13 and each delegate loudspeaker 5 is connected to a plurality of loudspeaker channels 14 .
  • All microphone channels 13 and all loudspeaker channels 14 are connected with a control unit 15 , which allows a central audio processing for example in view of volume and tone control, equalizing, acoustical feedback, suppression and/or scrambling to hide the identity of the speaker (for example used in courtrooms) etc.
  • each active delegate microphone 8 for each active delegate microphone 8 one of the microphone channels 13 is used to transport the audio signals to the control unit 15 .
  • the same number of the loudspeaker channels 14 is used to transfer the audio signals from the control unit 15 to the delegate units 2 .
  • Each delegate unit 2 is connected to each of the active loudspeaker channels 14 in order to receive the audio signals resulting from the active delegate microphones 8 .
  • the delegate unit 2 comprises a delay unit 16 , which is operable and/or adapted to add an individual time delay to each of the audio signals. The individual time delay is dependent on the distance between the respective delegate unit 2 and the active microphone 8 of the respective audio signal.
  • time delays d 21 , d 22 , and d 23 are added to the delegate unit 10 .
  • individual time delays d 11 , d 12 , d 13 and d 31 , d 32 , d 33 are added to the delegate units 9 and 11 , respectively.
  • the length of the time delays d 11 to d 33 is estimated by the delay unit 16 , for example on basis of an encoded position stamp in the audio signals, on basis of the selection of the loudspeaker channel 14 , etc.
  • the microphone channels 13 and the loudspeaker channels 14 are realized as an audio data stream channel, whereby the audio signals are digital or analog represented.
  • FIG. 3 shows a schematic view of a circular buffer 17 , which is employed in the delay unit 16 .
  • the circular buffer 17 shows a plurality of memory locations 0, . . . (N ⁇ 1) arranged in a circular shape.
  • the shape does only represent the architecture of the circular buffer 17 , the physical representation may be arranged in another way, for example in rows.
  • a write pointer W and a read pointer R are used to write and read, respectively, samples of the audio input signal in the memory locations. Both pointers W, R are moving in a clockwise direction so that the write pointer W writes samples 51 from the input audio signal to the circular buffer 17 and the read pointer R reads these segments 51 from the circular buffer 17 . After each writing or reading step the pointers R, W are moved to the next memory location. Furthermore, after reading a memory location this memory location is set to zero.
  • the distance between the write pointer W and the read pointer R determines the time delay d old , generated by the delay unit 16 .
  • the audio input signal source changes, for example a new speaker starts to speak
  • a changeover from the first audio input signal to the second audio input signal must be performed.
  • processing the second audio input signal may require another time delay, as the new speaker may be situated at another position as the first speaker.
  • FIG. 4 illustrates the case when the time delay of the first audio input signal D old , is the same as the time delay D new , of the second audio input signal.
  • the write pointer W finishes writing samples S 1 of the first audio input signal and starts to write down samples S 2 of the second audio input signal. The changeover will be performed without involving any problems.
  • FIG. 5 shows the case, when the time delay of the second audio input signal is longer than the time delay of the first audio input signal.
  • the write pointer W is moved from its old position to a new position, which is determined by the new time delay of the second audio input signal.
  • the read pointer erases the samples after reading the circular buffer 17 is filled with samples S 1 of the first input audio signal, then with a plurality of zeros and then with samples S 2 of the second audio input signal.
  • the first audio input signal will stop, a short period of silence will follow and then the second audio input signal will start.
  • FIG. 6 shows the case when the time delay of the second audio input signal is shorter than the time delay of the first audio input signal.
  • the write pointer W will be moved counter-clockwise and will arrive at a memory location, which is already filled with a sample S 1 of the first audio input signal.
  • the delay unit 16 or the write pointer W will write the sample S 2 additionally into the memory location, so that both audio input signals will overlap for some time.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)
US13/395,147 2009-09-03 2009-09-03 Delay unit for a conference audio system, method for delaying audio input signals, computer program and conference audio system Active 2031-12-19 US9271096B2 (en)

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US10334360B2 (en) * 2017-06-12 2019-06-25 Revolabs, Inc Method for accurately calculating the direction of arrival of sound at a microphone array
CN110096250B (zh) * 2018-01-31 2020-05-29 北京金山云网络技术有限公司 一种音频数据处理方法、装置、电子设备及存储介质
CN108897700A (zh) * 2018-06-26 2018-11-27 青岛海信宽带多媒体技术有限公司 一种环形缓存器的数据处理方法、装置及机顶盒
CN117880696B (zh) * 2022-10-12 2024-07-16 广州开得联软件技术有限公司 混音方法、装置、计算机设备以及存储介质

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JP5511961B2 (ja) 2014-06-04
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