WO2022161446A1 - 控制方法、装置和电子设备 - Google Patents

控制方法、装置和电子设备 Download PDF

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Publication number
WO2022161446A1
WO2022161446A1 PCT/CN2022/074425 CN2022074425W WO2022161446A1 WO 2022161446 A1 WO2022161446 A1 WO 2022161446A1 CN 2022074425 W CN2022074425 W CN 2022074425W WO 2022161446 A1 WO2022161446 A1 WO 2022161446A1
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Prior art keywords
sound
sound signal
preset
playback
preset range
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PCT/CN2022/074425
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English (en)
French (fr)
Inventor
周泽
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维沃移动通信有限公司
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Publication of WO2022161446A1 publication Critical patent/WO2022161446A1/zh

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/12Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R29/00Monitoring arrangements; Testing arrangements
    • H04R29/001Monitoring arrangements; Testing arrangements for loudspeakers

Definitions

  • the present application belongs to the field of electronic technology, and specifically relates to a control method, device and electronic device.
  • a microphone and a speaker are provided in the venue, so that the content of the narrator can be transmitted to the audience at the venue through the pickup function of the microphone and the amplification function of the speaker.
  • the content of the narrator cannot be clearly conveyed to every audience at the venue. For example, a listener sitting in a corner may not be able to hear what the narrator is saying because of the distance from the speakers.
  • the inventor found at least the following problems in the prior art: in the scene of an offline meeting, it is impossible to ensure that the content narrated by the narrator can be clearly delivered to every audience at the meeting.
  • the purpose of the embodiments of the present application is to provide a control method, which can solve the problem of inability to ensure that the content narrated by the narrator can be clearly transmitted to every audience at the venue in the scenario of an offline venue speech.
  • an embodiment of the present application provides a control method, the method includes: identifying at least two sound playback devices and at least two sound pickup devices in a target network; playing a first sound signal on the first sound playback device Under the circumstance, if the first sound signal received at the first position does not meet the first preset condition, the second sound playback device within the first preset range is controlled to play the first sound signal; A position is associated; when the first sound pickup device corresponding to the second position picks up the second sound signal, if the second sound signal does not meet the second preset condition, control the second sound within the second preset range. The pickup device picks up the second sound signal; the second preset range is associated with the second position.
  • an embodiment of the present application provides a control device, the device includes: an identification module for identifying at least two sound playback devices and at least two sound pickup devices in a target network; a playback compensation module for When the first sound playing device plays the first sound signal, if the first sound signal received at the first position does not meet the first preset condition, the second sound playing device within the first preset range is controlled to play the first sound signal.
  • the first preset range is associated with the first position; the sound pickup compensation module is used for picking up the second sound signal by the first sound pickup device corresponding to the second position, if the second sound signal does not satisfy the For the second preset condition, the second sound pickup device within the second preset range is controlled to pick up the second sound signal; the second preset range is associated with the second position.
  • an embodiment of the present application provides an electronic device, the electronic device includes a processor, a memory, and a program or instruction stored in the memory and executable on the processor.
  • the program or instruction is executed by the processor, the The steps of the method of the first aspect.
  • an embodiment of the present application provides a readable storage medium, where a program or an instruction is stored on the readable storage medium, and when the program or instruction is executed by a processor, the steps of the method of the first aspect are implemented.
  • an embodiment of the present application provides a chip, where the chip includes a processor and a communication interface, the communication interface is coupled to the processor, and the processor is configured to run programs or instructions to implement the method of the first aspect.
  • At least two sound playback devices and at least two sound pickup devices are located in the target network, so that in the target network, unified management and control can be performed for all sound playback devices and sound pickup devices.
  • the first sound playing device plays the first sound signal as a public device
  • the received first sound signal does not meet the first preset condition, such as making the first sound signal
  • the first preset range can be determined at the first position, and then another second sound playing device can be called in the first preset range, which is the same as the first sound.
  • the playing device plays the first sound signal synchronously, so that the user at the first position can hear the first sound signal clearly.
  • the first sound pickup device picks up the second sound signal
  • the picked up second sound signal does not meet the second preset condition, for example, it cannot be transmitted to every listener after being played, so that the second sound signal can be picked up when the second sound signal is picked up.
  • a second preset range is determined, and then in the second preset range, other second sound pickup devices are called to pick up the second sound signal synchronously with the first sound pickup device, so that the first sound pickup device can pick up the second sound signal.
  • the two sound signals After the two sound signals are played, they can be transmitted to every listener. It can be seen that this embodiment ensures that the content narrated by the narrator in the venue can be clearly transmitted to every listener in the venue from the above-mentioned two aspects of sound playback and sound pickup.
  • Fig. 2 is one of the schematic diagrams of the device distribution in the venue provided by the embodiment of the present application;
  • FIG 3 is the second schematic diagram of the distribution of devices in the venue provided by the embodiment of the present application.
  • FIG. 4 is a block diagram of a control device provided by an embodiment of the present application.
  • FIG. 5 is a schematic diagram of a hardware structure of an electronic device provided by an embodiment of the present application.
  • FIG. 6 is a second schematic diagram of a hardware structure of an electronic device provided by an embodiment of the present application.
  • first, second and the like in the description and claims of the present application are used to distinguish similar objects, and are not used to describe a specific order or sequence. It is to be understood that the data so used are interchangeable under appropriate circumstances so that the embodiments of the present application can be practiced in sequences other than those illustrated or described herein, and distinguish between “first”, “second”, etc.
  • the objects are usually of one type, and the number of objects is not limited.
  • the first object may be one or more than one.
  • “and/or” in the description and the claims indicates at least one of the connected objects, and the character “/" generally indicates that the associated objects are in an "or” relationship.
  • FIG. 1 shows a flowchart of a control method provided by an embodiment of the present application, and the method includes:
  • S1 Identify at least two sound playback devices and at least two sound pickup devices in the target network.
  • the target network is used to represent the same network.
  • the connection can be automatically added or released in time.
  • WIFI wireless fidelity
  • P2P wireless fidelity
  • the sound playing device includes a public sound installed in the venue.
  • the sound playing apparatus further includes the personal device in the venue, such as a mobile phone.
  • the sound pickup device includes a public microphone in the venue. Further, if the user's personal device includes a microphone, the sound pickup device also includes the personal device in the venue, such as a mobile phone.
  • each personal device it can be used as both the sound playing device in this embodiment and the sound pickup device in this embodiment.
  • the user can manually connect the personal device to the target network.
  • the first preset range is associated with the first position.
  • the number of sound playing devices is at least two.
  • the number of the first sound playing devices and the number of the second sound playing devices are not limited.
  • the first sound playing device is a public sound in the venue, and is used to play narration content, teaching content, speech content, and the like.
  • the second sound playing device is a personal device in the venue, and is used to assist the first sound playing device to complete playing.
  • the first sound signal played at the sound quality blind spot does not meet the first preset condition.
  • the inaudible first sound signal is used as the first sound signal that does not meet the first preset condition.
  • the set threshold may be associated with the actual user experience.
  • the first position is used to represent a sound quality blind spot.
  • the first preset range is determined around the first position.
  • the second sound playback device within the first preset range is controlled to play the first sound signal together with the first sound playback device, thereby improving the first sound signal received at the first position so that it satisfies the first preset condition. In this way, the content of the narrator can be clearly conveyed to every audience at the venue.
  • the first preset range includes the first position.
  • Step S3 in the case where the first sound pickup device corresponding to the second position picks up the second sound signal, if the second sound signal does not meet the second preset condition, then control the second sound pickup device within the second preset range Pick up the second sound signal.
  • the second preset range is associated with the second position.
  • the number of sound pickup devices is at least two.
  • the number of the first sound pickup device and the number of the second sound pickup device are not limited.
  • the first sound pickup device may be a public microphone in the venue, or a personal device in the venue, or both.
  • the first sound pickup device is used to pick up narration content, lecture content, speech content, and the like.
  • the second sound playing device is a personal device in the conference venue.
  • the second sound playing device is used for assisting the first sound pickup device to complete the pickup.
  • the application scenario involved in this step is, for example, when the narrator uses a microphone to narrate on the stage, the microphone acts as the first sound pickup device to pick up the narration content, and correspondingly, the second sound signal includes the narration content.
  • Another example of the application scenario involved in this step is that when a speaker speaks at the audience seat, the microphone or the speaker's mobile phone is used as the first sound pickup device to pick up the speech content, and correspondingly, the second sound signal includes the speech content.
  • the second sound signal picked up by the first sound pickup device can be obtained to determine whether the second sound signal satisfies the second preset condition.
  • the second sound signal does not meet the second preset condition.
  • the acquired second sound signal is played through all the playback devices in the venue, it cannot be ensured that every listener in the venue can hear clearly, and it is considered that the second sound signal does not meet the second preset condition.
  • the set threshold may be associated with the actual playback effect.
  • the second position is used to represent the emission position of the second sound signal.
  • the second sound pickup device in the second preset range is controlled to pick up the second sound signal together with the first sound pickup device, thereby improving the second sound signal picked up at the second position so that it satisfies the second preset condition .
  • the phenomenon of poor playback effect due to poor sound pickup can be improved, so that the content of the narrator can be clearly transmitted to the audience at the venue. every listener.
  • At least two sound playback devices and at least two sound pickup devices are located in the target network, so that in the target network, unified management and control can be performed for all sound playback devices and sound pickup devices.
  • the first sound playing device plays the first sound signal as a public device
  • the received first sound signal does not meet the first preset condition, such as making the first sound signal
  • the first preset range can be determined at the first position, and then another second sound playing device can be called in the first preset range, which is the same as the first sound.
  • the playing device plays the first sound signal synchronously, so that the user at the first position can hear the first sound signal clearly.
  • the first sound pickup device picks up the second sound signal
  • the second sound signal can be picked up when the second sound signal is picked up.
  • a second preset range is determined, and then in the second preset range, other second sound pickup devices are called to pick up the second sound signal synchronously with the first sound pickup device, so that the first sound pickup device can pick up the second sound signal.
  • the two sound signals After the two sound signals are played, they can be transmitted to every listener. It can be seen that this embodiment ensures that the content narrated by the narrator in the venue can be clearly transmitted to every listener in the venue from the above-mentioned two aspects of sound playback and sound pickup.
  • the first preset condition includes a first preset sound level. Accordingly, S2 may include steps A1, A2 and A3:
  • A1 Determine the first sound compensation amount according to the first sound level of the first sound signal received at the first position and the first preset sound level.
  • the sound may be divided into multiple levels according to the different degrees of the sound quality and intensity of the sound.
  • the first preset sound level is used to indicate a sound level that can reach a certain sound quality and a certain intensity, and the sound signal at the sound level can ensure that the user can hear clearly. Further, the sound signal at this sound level can ensure that the user can hear high-quality sound.
  • the first sound level of the first sound signal at the first position may be acquired to determine the sound quality and intensity of the first sound signal at the first position.
  • the first sound compensation amount may be determined based on the first sound level and the first preset sound level by analyzing the lack level.
  • the first sound compensation amount is equal to the magnitude of the difference between the first sound level and the first preset sound level.
  • A2 Determine a first preset range with the first position as the center and the first distance as the radius according to at least one of the first sound compensation amount and the density of the second sound playback devices around the first position.
  • the radius of the compensation range is defined by the magnitude of the deficiency, wherein the radius of the compensation range is the first distance.
  • the first distance is larger, so that more users' devices can be within the first preset range; otherwise, if the magnitude of the deficiency is smaller, the first distance is smaller.
  • the first distance is also related to the distribution of devices around the first location. For example, if the devices around the first location are densely distributed, the first distance is relatively small; on the contrary, if the devices around the first location are scattered, the first distance is relatively large, so that more users' devices can be located in the first location. within the preset range.
  • the setting criteria of the first preset sound level may include at least one of a level of high-quality sound and a level of basic sound.
  • a level of high-quality sound and a level of basic sound.
  • the first preset range try to make the first sound signal at the first position reach the level of high-quality sound; and combine the actual situation of the first position, such as the corner, and the sound around the first position If the distribution of the playback device is sparse, at least the first sound signal at the first position can reach the level of the basic sound.
  • the location where the target audience is located is the first location
  • the circle where the first location is located is the determined first preset range.
  • the other listeners shown in FIG. 2 are other listeners in the venue except the assisting listeners and the target listeners.
  • A3 According to the number of second sound playing devices on each radius of the first preset range, and the distance between each second sound playing device and the first position, determine the number of times each second sound playing device plays the first sound signal Playback parameters.
  • the playback parameters at least include playback volume and playback direction.
  • each second sound playing device provides compensation sounds with different volumes and sounding directions according to the positional distance from the first position. In this way, combined with the comprehensive sound of the compensation sound and the public sound of the venue, a timely and repeated sound quality evaluation is carried out to ensure that the target audience in the first position is in the high-quality sound effect range.
  • the fewer devices are compensated on a single radius, the more distant devices will be compensated more strongly, keeping the sound around the listener's ears balanced.
  • the sound quality compensation scheme is determined in real time according to the analysis result of the sound quality analysis module in real time.
  • the sound quality compensation scheme includes at least a compensation radius and playback parameters of each second sound playback device on the compensation radius.
  • a specific sound quality compensation scheme is provided for the sound quality blind spot, so that the target audience of the sound quality blind spot in the venue can be in the high-quality sound effect range, so as to ensure the content of the narrator in the venue. It can be clearly conveyed to every audience in the venue.
  • the second preset condition includes a second preset sound level. Accordingly, S3 may include steps B1 and B2:
  • B1 Determine the second sound compensation amount according to the second sound level of the second sound signal picked up by the first sound pickup device and the second preset sound level.
  • the second preset sound level is used to indicate a sound level that can reach a certain sound quality and a certain intensity, and the sound signal under this sound level can be played to ensure that users in the venue can hear it clearly. Furthermore, the sound signal at this sound level can ensure that the user can hear high-quality sound after being played.
  • the second sound compensation amount may be determined based on the second sound level and the second preset sound level by analyzing the lack level.
  • the second sound compensation amount is equal to the magnitude of the difference between the second sound level and the second preset sound level.
  • B2 Determine a second preset range with the second position as the center and the second distance as the radius according to at least one of the second sound compensation amount and the density of the second sound pickup devices around the second position.
  • the radius of the compensation range that is, the second distance, is delineated by the lack of magnitude.
  • the second distance is also related to the distribution of devices around the second location. For example, if the devices around the second location are densely distributed, the second distance is relatively small; conversely, if the devices around the second location are scattered, the second distance is relatively large, so that more users' devices can be located in the second location. within the preset range.
  • the setting criteria of the second preset sound level may include: a level of high-quality sound and a level of basic sound.
  • the second sound signal when determining the second preset range, the second sound signal can be played as far as possible to achieve a high-quality sound level; however, in combination with the actual situation of the second location, for example, the distribution of sound pickup devices around the second location is sparse. , then at least the second sound signal can reach the level of the basic sound after playing.
  • this embodiment is more suitable for an interactive session, where the audience speaks.
  • the compensation cluster (the second sound pickup device) is determined according to the lack of the sound source of the compensation object (the first sound pickup device), and dynamically Change the compensation parameters, and use the mobile phone or microphone around the compensation object to collect sound to form a better sound collection effect. Further, dynamic adjustment is performed according to changes in the surrounding scene.
  • the quality evaluation of the point-to-point audio source is performed, and the magnitude of the respective lack thereof is determined. Since the density of other listeners around the two locations is different, the location distance radius and compensation parameters are set differently (in the figure on the left side of the speaker, there are relatively few users around the speaker, and the compensation radius is expanded).
  • dynamic compensation is performed to assist the switching of the audience circle.
  • the other listeners shown in FIG. 3 are other listeners in the venue except the assisting listeners.
  • the compensation parameter includes the second distance.
  • a specific sound quality compensation scheme for poor pickup effect is provided.
  • it has a more intelligent sound collection tendency. It can start the local equipment according to the position of the user's voice, and compensate the speaker according to the sound intensity and quality of the peripheral equipment, so that the content of the speech can be clearly transmitted to every participant in the venue during the interactive session. a listener.
  • this embodiment has a more intelligent human voice analysis capability. When multiple people speak alternately, the sound source can be automatically distinguished according to the volume and timbre, and peripheral devices can be activated for dynamic sound collection.
  • the sound pickup method of this embodiment is more convenient and intelligent, so as to achieve the effects of improving the enthusiasm of the audience to speak, reducing the cost, and improving the speaking efficiency. Make the venue more upscale.
  • C1 Determine the third sound level of the first sound signal.
  • the third sound level is a sound level attainable by the first sound signal, and the sound level has a certain sound quality, intensity, and the like.
  • C3 Determine, according to the third sound level and the third location, whether the first sound signals received at each location in the target network satisfy the first preset condition.
  • the third position of the first sound playing device can show the sound distribution in the venue, so that combined with the third sound level of the picked up first sound signal itself, the quality analysis of the sound effect structure of the whole venue can be performed.
  • each location in the target network includes a node that joins the cluster, that is, the location of the personal device in the conference venue.
  • the first sound signal played by the first sound playing device has better sound quality and higher intensity, and the public sound in the venue is evenly distributed in every corner, so that every position in the venue can be heard clearly.
  • the intensity of the first sound signal played by the first sound playback device is relatively low. Even if the public audio in the venue is evenly distributed in every corner, there will still be sound quality blind spots in the venue.
  • the public sound in the venue is not uniform, and the intensity of the first sound signal played by the first sound playback device is relatively low, so the venue will inevitably have a sound quality blind spot.
  • a local audio cluster is first established in combination with the audience's mobile phone resources, and then a comprehensive analysis is carried out in combination with the audience node location and the overall sound effect structure quality, and a sound receiving point model is initially established, so as to quickly and accurately identify sound quality blind spots , and then make full use of the existing resources, dynamically evaluate the playback quality and adjust the compensation scheme, and perform sound compensation for the sound quality blind spots, so as to solve the problem of lack of sound effects in the corner positions and other blind spots of sound quality.
  • step D1 In the flow of the control method according to another embodiment of the present application, before S3, it further includes step D1:
  • D1 At the second position, when the decibel value of the human voice sound signal is greater than the decibel value of the ambient sound signal, the first sound pickup device is controlled to pick up the second sound signal.
  • the second sound signal includes an ambient sound signal and a human voice sound signal.
  • the decibel value of the human voice signal is greater than the decibel value of the ambient sound signal, it is considered that someone is speaking at that position, and the first sound pickup device at this position is controlled to pick up the first sound. Two sound signals.
  • One way is: according to the second position, the corresponding first sound pickup device is identified, and the microphone function of the first sound pickup device is controlled to be turned on; the other way is: when the human voice sound signal reaches a certain preset decibel value, The microphone function of the first sound pickup device at the corresponding position is automatically activated.
  • the number of the second positions includes a plurality. For example, in a speaking scenario, listeners at multiple locations alternate speaking.
  • different second sound signals can be distinguished by characteristics such as timbre.
  • this embodiment can also perform sound analysis and compensation for different speakers at different positions in combination with characteristic information such as the timbre of the second sound signal at different second positions.
  • a method for detecting the second position is provided.
  • the basic venue sound source analysis will be performed to determine the speaker position, sound source quality, and sound source quantity, so as to quickly and accurately identify the speaking position, so as to perform sound analysis and compensation for the speaking position, and then combine the audience's mobile phone resources.
  • Establish a local microphone cluster make full use of existing resources, and dynamically perform sound quality evaluation and compensation plan adjustment, reducing the labor cost of microphone equipment and transmission.
  • the sound source and sound transmission are intelligently analyzed and collected, and the compensation operation is performed for the sound transmission and sound collection at the venue, which improves the listening quality of the audience at the venue and the interaction of questions in the speech scene. Efficiency and positivity.
  • both the sound of the venue and the mobile phone of the audience can be used as the sound, and the uniformity adjustment can be performed according to the distribution of the sound in the venue, so as to provide a more uniform sound cluster;
  • the listener's microphone and the audience's mobile phone are in the same local area network environment, and the device can record the audio source, forming a large number of evenly distributed microphone clusters, providing a more convenient connection method and a uniform microphone cluster.
  • the intelligent and timely quality assessment and resource mobilization scheme design of the embodiments of the present application closely connect the entire venue with the actual equipment and the surrounding environment of personnel, and the application scope and ability are extremely strong ;
  • the trouble of passing the microphone is generally solved, so that the whole audience can participate in the interaction with high quality.
  • the intelligent resource mobilization scheme of the cluster equipment can utilize the design of the urban ambient light source to dynamically adjust the light source calling object and corresponding parameters according to the actual quality and demand.
  • the execution body may be a control device, or a control module in the control device for executing the control method.
  • the control device of the control method provided by the embodiment of the present application is described by taking the control device executing the control method as an example.
  • FIG. 4 shows a block diagram of a control device according to another embodiment of the present application, including:
  • an identification module 10 for identifying at least two sound playback devices and at least two sound pickup devices in the target network
  • the playback compensation module 20 is configured to control the sound signal within the first predetermined range if the first sound signal received at the first position does not meet the first preset condition when the first sound playback device plays the first sound signal.
  • the second sound playing device plays the first sound signal; the first preset range is associated with the first position;
  • the sound pickup compensation module 30 is configured to control the sound pickup within the second preset range if the second sound signal does not meet the second preset condition when the first sound pickup device corresponding to the second position picks up the second sound signal.
  • the second sound pickup device picks up the second sound signal; the second preset range is associated with the second position.
  • At least two sound playback devices and at least two sound pickup devices are located in the target network, so that in the target network, unified management and control can be performed for all sound playback devices and sound pickup devices .
  • the first sound playing device plays the first sound signal as a public device
  • the received first sound signal does not meet the first preset condition, such as making the first sound signal
  • the first preset range can be determined at the first position, and then another second sound playback device can be called in the first preset range, which is the same as the first sound.
  • the playback device plays the first sound signal synchronously, so that the user at the first position can hear the first sound signal clearly.
  • the first sound pickup device picks up the second sound signal
  • the picked up second sound signal does not meet the second preset condition, for example, it cannot be transmitted to every listener after being played, so that the second sound signal can be picked up when the second sound signal is picked up.
  • a second preset range is determined, and then in the second preset range, other second sound pickup devices are called to pick up the second sound signal synchronously with the first sound pickup device, so that the first sound pickup device can pick up the second sound signal.
  • the two sound signals After the two sound signals are played, they can be transmitted to every listener. It can be seen that this embodiment ensures that the content narrated by the narrator in the venue can be clearly transmitted to every listener in the venue from the above-mentioned two aspects of sound playback and sound pickup.
  • the first preset condition includes a first preset sound level
  • the playback compensation module 20 includes:
  • a first determining unit configured to determine the first sound compensation amount according to the first sound level of the first sound signal received at the first position and the first preset sound level
  • the second determining unit is configured to determine, according to at least one of the first sound compensation amount and the density of the second sound playing devices around the first position, a first preset with the first position as the center and the first distance as the radius set range;
  • a third determining unit configured to determine the playback of each second sound playback device according to the number of the second sound playback devices on each radius of the first preset range and the distance between each second sound playback device and the first position Playing parameters of the first sound signal; the playing parameters include at least the playing volume and the playing direction.
  • the second preset condition includes a second preset sound level
  • the pickup compensation module 30 includes:
  • a fourth determining unit configured to determine the second sound compensation amount according to the second sound level of the second sound signal picked up by the first sound pickup device and the second preset sound level
  • the fifth determination unit is configured to determine a second preset with the second position as the center and the second distance as the radius according to at least one of the second sound compensation amount and the density of the second sound pickup devices around the second position. set range.
  • the device further includes:
  • a sixth determination module used for determining the third sound level of the first sound signal
  • an acquisition module for acquiring the third position of the first sound playback device
  • the seventh determination module is configured to determine, according to the third sound level and the third position, whether the first sound signals received at each position in the target network satisfy the first preset condition.
  • the device further includes:
  • the pickup module is configured to control the first sound pickup device to pick up the second sound signal when the decibel value of the human voice sound signal is greater than the decibel value of the ambient sound signal at the second position; the second sound signal includes the ambient sound signal and Human voice sound signal.
  • the control device in this embodiment of the present application may be a device, or may be a component, an integrated circuit, or a chip in a terminal.
  • the apparatus may be a mobile electronic device or a non-mobile electronic device.
  • the mobile electronic device may be a mobile phone, a tablet computer, a notebook computer, a palmtop computer, an in-vehicle electronic device, a wearable device, an ultra-mobile personal computer (UMPC), a netbook, or a personal digital assistant (personal digital assistant).
  • UMPC ultra-mobile personal computer
  • PDA personal digital assistant
  • non-mobile electronic devices can be servers, network attached storage (Network Attached Storage, NAS), personal computer (personal computer, PC), television (television, TV), teller machine or self-service machine, etc., this application Examples are not specifically limited.
  • the control device in this embodiment of the present application may be a device with an operating system.
  • the operating system may be an Android (Android) operating system, an ios operating system, or other possible operating systems, which are not specifically limited in the embodiments of the present application.
  • control device provided in the embodiment of the present application can implement each process implemented by the foregoing method embodiment, which is not repeated here to avoid repetition.
  • an embodiment of the present application further provides an electronic device 100, including a processor 101, a memory 102, a program or instruction stored in the memory 102 and executable on the processor 101,
  • an electronic device 100 including a processor 101, a memory 102, a program or instruction stored in the memory 102 and executable on the processor 101,
  • the program or instruction is executed by the processor 101, each process of any one of the above control method embodiments can be implemented, and the same technical effect can be achieved. To avoid repetition, details are not repeated here.
  • the electronic devices in the embodiments of the present application include the aforementioned mobile electronic devices and non-mobile electronic devices.
  • FIG. 6 is a schematic diagram of a hardware structure of an electronic device implementing an embodiment of the present application.
  • the electronic device 1000 includes but is not limited to: a radio frequency unit 1001, a network module 1002, an audio output unit 1003, an input unit 1004, a sensor 1005, a display unit 1006, a user input unit 1007, an interface unit 1008, a memory 1009, a processor 1010 and other components .
  • the electronic device 1000 may also include a power source (such as a battery) for supplying power to various components, and the power source may be logically connected to the processor 1010 through a power management system, so that the power management system can manage charging, discharging, and power functions. consumption management and other functions.
  • a power source such as a battery
  • the power management system can manage charging, discharging, and power functions. consumption management and other functions.
  • the structure of the electronic device shown in FIG. 6 does not constitute a limitation on the electronic device, and the electronic device may include more or less components than those shown in the figure, or combine some components, or arrange different components, which will not be repeated here. .
  • the processor 1010 is used to identify at least two sound playback devices and at least two sound pickup devices in the target network; when the first sound playback device plays the first sound signal, if the If the first sound signal does not meet the first preset condition, the second sound playback device within the first preset range is controlled to play the first sound signal; the first preset range is associated with the first position; When the first sound pickup device picks up the second sound signal, if the second sound signal does not meet the second preset condition, the second sound pickup device within the second preset range is controlled to pick up the second sound signal; The set range is associated with the second location.
  • At least two sound playback devices and at least two sound pickup devices are located in the target network, so that in the target network, unified management and control can be performed for all sound playback devices and sound pickup devices .
  • the first sound playing device plays the first sound signal as a public device
  • the received first sound signal does not meet the first preset condition, such as making the first sound signal
  • the first preset range can be determined at the first position, and then another second sound playing device can be called in the first preset range, which is the same as the first sound.
  • the playing device plays the first sound signal synchronously, so that the user at the first position can hear the first sound signal clearly.
  • the first sound pickup device picks up the second sound signal
  • the picked up second sound signal does not meet the second preset condition, for example, it cannot be transmitted to every listener after being played, so that the second sound signal can be picked up when the second sound signal is picked up.
  • a second preset range is determined, and then in the second preset range, other second sound pickup devices are called to pick up the second sound signal synchronously with the first sound pickup device, so that the first sound pickup device can pick up the second sound signal.
  • the two sound signals After the two sound signals are played, they can be transmitted to every listener. It can be seen that this embodiment ensures that the content narrated by the narrator in the venue can be clearly transmitted to every listener in the venue from the above-mentioned two aspects of sound playback and sound pickup.
  • the first preset condition includes a first preset sound level
  • the processor 1010 is further configured to determine according to the first sound level of the first sound signal received at the first position and the first preset sound level a first sound compensation amount; according to at least one of the first sound compensation amount and the density of the second sound playback devices around the first position, determine a first preset with the first position as the center and the first distance as the radius range; according to the number of second sound playback devices on each radius of the first preset range, and the distance between each second sound playback device and the first position, determine the size of each second sound playback device for playing the first sound signal Playback parameters; playback parameters at least include playback volume and playback direction.
  • the second preset condition includes a second preset sound level
  • the processor 1010 is further configured to, according to the second sound level of the second sound signal picked up by the first sound pickup device, and the second preset sound level , determine the second sound compensation amount; according to at least one of the second sound compensation amount and the density of the second sound pickup devices around the second position, determine the second position with the second position as the center and the second distance as the radius. Preset range.
  • the processor 1010 is further configured to determine a third sound level of the first sound signal; obtain a third position of the first sound playback device; determine each position in the target network according to the third sound level and the third position Whether the received first sound signal satisfies the first preset condition.
  • the processor 1010 is further configured to control the first sound pickup device to pick up the second sound signal when the decibel value of the human voice sound signal is greater than the decibel value of the ambient sound signal at the second position; the second sound The signals include ambient sound signals and human voice sound signals.
  • the sound source and sound transmission are intelligently analyzed and collected, and the compensation operation is performed for the sound transmission and sound collection at the venue, which improves the listening quality of the audience at the venue and the interaction of questions in the speech scene. Efficiency and positivity.
  • both the sound of the venue and the mobile phone of the audience can be used as the sound, and the uniformity adjustment can be performed according to the distribution of the sound in the venue, so as to provide a more uniform sound cluster;
  • the listener's microphone and the audience's mobile phone are in the same local area network environment, and the device can record the audio source, forming a large number of evenly distributed microphone clusters, providing a more convenient connection method and a uniform microphone cluster.
  • the intelligent and timely quality assessment and resource mobilization scheme design of the embodiments of the present application closely connect the entire venue with the actual equipment and the surrounding environment of personnel, and the application scope and ability are extremely strong ;
  • the trouble of passing the microphone is generally solved, so that the whole audience can participate in the interaction with high quality.
  • the input unit 1004 may include a graphics processor (Graphics Processing Unit, GPU) 10041 and a microphone 10042. Such as camera) to obtain still pictures or video image data for processing.
  • the display unit 1006 may include a display panel 10061, which may be configured in the form of a liquid crystal display, an organic light emitting diode, or the like.
  • the user input unit 1007 includes a touch panel 10071 and other input devices 10072 .
  • the touch panel 10071 is also called a touch screen.
  • the touch panel 10071 may include two parts, a touch detection device and a touch controller.
  • Other input devices 10072 may include, but are not limited to, physical keyboards, function keys (such as volume control keys, switch keys, etc.), trackballs, mice, and joysticks, which will not be repeated here.
  • Memory 1009 may be used to store software programs as well as various data, including but not limited to application programs and operating systems.
  • the processor 1010 may integrate an application processor and a modem processor, wherein the application processor mainly processes the operating system, user interface, and application programs, and the like, and the modem processor mainly processes wireless communication. It can be understood that, the above-mentioned modulation and demodulation processor may not be integrated into the processor 1010.
  • Embodiments of the present application further provide a readable storage medium, where a program or an instruction is stored on the readable storage medium. When the program or instruction is executed by a processor, each process of any of the above control method embodiments can be implemented, and the same can be achieved. In order to avoid repetition, the technical effect will not be repeated here.
  • the processor is the processor in the electronic device in the above embodiment.
  • readable storage media include computer-readable storage media, such as computer read-only memory (Read-Only Memory, ROM), random access memory (Random Access Memory, RAM), non-transitory computer-readable storage media such as magnetic or optical disks. Read the storage medium.
  • the embodiment of the present application further provides a chip, the chip includes a processor and a communication interface, the communication interface and the processor are coupled, and the processor is used for running a program or an instruction to implement each process of any of the above control method embodiments, and can achieve The same technical effect, in order to avoid repetition, will not be repeated here.
  • the chip mentioned in the embodiments of the present application may also be referred to as a system-on-chip, a system-on-chip, a system-on-a-chip, or a system-on-a-chip, or the like.
  • the method of the above embodiment can be implemented by means of software plus a necessary general hardware platform, and of course can also be implemented by hardware, but in many cases the former is better implementation.
  • the technical solution of the present application can be embodied in the form of a software product in essence or in a part that contributes to the prior art, and the computer software product is stored in a storage medium (such as ROM/RAM, magnetic disk, CD-ROM), including several instructions to make a terminal (which may be a mobile phone, a computer, a server, an air conditioner, or a network device, etc.) execute the methods described in the various embodiments of this application.
  • a storage medium such as ROM/RAM, magnetic disk, CD-ROM

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Abstract

本申请的实施例提供了一种控制方法、装置及电子设备,其中,该方法包括:识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第一预设条件,则控制与该位置关联的第一预设范围内的第二声音播放装置播放第一声音信号;在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制与该位置关联的第二预设范围内的第二声音拾取装置拾取第二声音信号。本申请中的控制方法应用于电子设备中。

Description

控制方法、装置和电子设备
相关申请的交叉引用
本申请主张2021年01月28日在中国提交的中国专利申请号202110121918.1的优先权,其全部内容通过引用包含于此。
技术领域
本申请属于电子技术领域,具体涉及一种控制方法、装置和电子设备。
背景技术
目前,线下集中式会场演讲、授课越来越流行。通常,在会场中,设置有话筒和音箱,以便于讲述者讲述的内容可通过话筒的拾音功能,以及音箱的扩音功能,传递给会场的听众。
由于话筒和音箱等设备的限制,在一些情况下,使得讲述者讲述的内容并不能清晰地传递至会场的每一位听众。例如,坐在角落里的听众,可能会因为距离音箱较远,无法听清楚讲述者讲述的内容。
因此,在实现本申请过程中,发明人发现现有技术中至少存在如下问题:在线下会场演讲的场景中,无法确保讲述者讲述的内容能够清晰地传递至会场的每一位听众。
发明内容
本申请实施例的目的是提供一种控制方法,能够解决在线下会场演讲的场景中,无法确保讲述者讲述的内容能够清晰地传递至会场的每一位听众的问题。
第一方面,本申请实施例提供了一种控制方法,该方法包括:识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第 一预设条件,则控制第一预设范围内的第二声音播放装置播放第一声音信号;第一预设范围与第一位置相关联;在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取第二声音信号;第二预设范围与第二位置相关联。
第二方面,本申请实施例提供了一种控制装置,该装置包括:识别模块,用于识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;播放补偿模块,用于在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放第一声音信号;第一预设范围与第一位置相关联;拾音补偿模块,用于在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取第二声音信号;第二预设范围与第二位置相关联。
第三方面,本申请实施例提供了一种电子设备,该电子设备包括处理器、存储器及存储在存储器上并可在处理器上运行的程序或指令,程序或指令被处理器执行时实现如第一方面的方法的步骤。
第四方面,本申请实施例提供了一种可读存储介质,可读存储介质上存储程序或指令,程序或指令被处理器执行时实现如第一方面的方法的步骤。
第五方面,本申请实施例提供了一种芯片,芯片包括处理器和通信接口,通信接口和处理器耦合,处理器用于运行程序或指令,实现如第一方面的方法。
在本申请的实施例中,至少两个声音播放装置和至少两个声音拾取装置处于目标网络中,从而在目标网络中,可针对全部的声音播放装置和声音拾取装置,进行统一管理控制。一方面,在第一声音播放装置作为公共设备播放第一声音信号的情况下,若检测出会场中的第一位置处,接收到的第一声音信号不满足第一预设条件,如使得第一位置处的用户听不清楚第一声音信号,从而可在第一位置处,确定第一预设范围,再在第一预设范围中,调用其它的第二声音播放装置,同第一声音播放装置同步播放第一声音信号,进而使得第一位置 处的用户可以听清楚第一声音信号。另一方面,在第一声音拾取装置拾取第二声音信号的情况下,若拾取的第二声音信号不满足第二预设条件,如播放后不能传递至每一位听众,从而可在拾取第二声音信号的第二位置处,确定第二预设范围,再在第二预设范围中,调用其它的第二声音拾取装置,同第一声音拾取装置同步拾取第二声音信号,进而使得第二声音信号播放后能够传递至每一位听众。可见,本实施例从上述声音播放和声音拾取两个方面,确保会场中的讲述者讲述的内容能够清晰地传递至会场的每一位听众。
附图说明
图1是本申请实施例提供的控制方法的流程图;
图2是本申请实施例提供的会场内的装置分布示意图之一;
图3是本申请实施例提供的会场内的装置分布示意图之二;
图4是本申请实施例提供的控制装置的框图;
图5是本申请实施例提供的电子设备的硬件结构示意图之一。
图6是本申请实施例提供的电子设备的硬件结构示意图之二。
具体实施方式
下面将结合本申请实施例中的附图,对本申请实施例中的技术方案进行清楚、完整地描述,显然,所描述的实施例是本申请一部分实施例,而不是全部的实施例。基于本申请中的实施例,本领域普通技术人员在没有作出创造性劳动前提下所获得的所有其他实施例,都属于本申请保护的范围。
本申请的说明书和权利要求书中的术语“第一”、“第二”等是用于区别类似的对象,而不用于描述特定的顺序或先后次序。应该理解这样使用的数据在适当情况下可以互换,以便本申请的实施例能够以除了在这里图示或描述的那些以外的顺序实施,且“第一”、“第二”等所区分的对象通常为一类,并不限定对象的个数,例如第一对象可以是一个,也可以是多个。此外,说明书以及权利要求中“和/或”表示所连接对象的至少其中之一,字符“/”,一般 表示前后关联对象是一种“或”的关系。
下面结合附图,通过具体地实施例及其应用场景对本申请实施例提供的控制方法进行详细地说明。
图1示出了本申请一个实施例提供的控制方法的流程图,该方法包括:
S1:识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置。
可选地,目标网络用于表示同一网络。
可参考地,基于演讲、授课所在的会场,进行基础网络设备搭建,以形成局域网,在局域网中添加无线保真(Wireless Fidelity,WIFI)直连(P2P)能力,从而可将会场中的话筒、音响,以及进入会场用户的设备进行自动连接,形成整个局域联动集群模块,同时可以根据会场人员流动,及时自动添加或者解除连接。
对应地,声音播放装置包括安装在会场内的公共音响。进一步地,若用户的个人设备中包括扬声器,则声音播放装置还包括会场内的个人设备,如手机。
对应地,声音拾取装置包括会场内的公共话筒。进一步地,若用户的个人设备中包括麦克风,则声音拾取装置还包括会场内的个人设备,如手机。
可见,对于每一个个人设备,既可以作为本实施例中的声音播放装置,又可以作为本实施例中的声音拾取装置。
可选地,用户可将个人设备手动接入目标网络中。
S2:在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放第一声音信号。
其中,第一预设范围与第一位置相关联。
在本实施例中,声音播放装置的数量为至少两个。对应地,第一声音播放装置的数量,以及第二声音播放装置的数量不作限定。
可选地,第一声音播放装置为会场内的公共音响,用于播放讲述内容、授课内容、发言内容等。
可选地,第二声音播放装置为会场内的个人设备,用于协助第一声音播放装置完成播放。
在该步骤中,基于会场内的第一声音播放装置的分布,以及全场音效结构质量,可检测出会场内的音质盲点。
其中,在音质盲点处播放的第一声音信号不满足第一预设条件。
可选地,基于用户的实际体验,将听得不清楚的第一声音信号作为不满足第一预设条件的第一声音信号。
具体地,当会场内某位置,接收到的第一声音信号的音质、强度等,无法达到设定阈值时,认为该位置接收到的第一声音信号不满足第一预设条件。其中,设定阈值可与用户实际体验关联。
在该步骤中,第一位置用于表示音质盲点。
因此,在会场中检测出第一位置后,在第一位置周围确定第一预设范围。从而,控制第一预设范围内的第二声音播放装置同第一声音播放装置一起,播放第一声音信号,进而改善第一位置处接收到的第一声音信号,使其满足第一预设条件。这样,使得讲述者讲述的内容能够清晰地传递至会场的每一位听众。
可选地,第一预设范围包括第一位置。
步骤S3:在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取第二声音信号。
其中,第二预设范围与第二位置相关联。
在本实施例中,声音拾取装置的数量为至少两个。对应地,第一声音拾取装置的数量,以及第二声音拾取装置的数量不作限定。
可选地,第一声音拾取装置可以是会场内的公共话筒,也可以是会场内的个人设备,还可以是二者均包括。第一声音拾取装置用于拾取讲述内容、授课内容、发言内容等。
可选地,第二声音播放装置为会场内的个人设备。第二声音播放装置用于协助第一声音拾取装置完成拾取。
该步骤中所涉及的应用场景如,讲述者在舞台上使用话筒讲述时,话筒作为第一声音拾取装置拾取讲述内容,对应地,第二声音信号包括讲述内容。
该步骤中所涉及的应用场景又如,发言者在听众位发言时,话筒或者发言者的手机作为第一声音拾取装置拾取发言内容,对应地,第二声音信号包括发言内容。
因此,在该步骤中的应用场景中,可获取第一声音拾取装置拾取的第二声音信号,以判断第二声音信号是否满足第二预设条件。
可选地,将获取的第二声音信号通过播放装置播放后,若不能达到预期的播放效果,认为第二声音信号不满足第二预设条件。
例如,将获取的第二声音信号通过会场内全部的播放装置播放后,也不能确保会场内的每一位听众均能听清楚,认为第二声音信号不满足第二预设条件。
具体地,第二声音信号的音质、强度等,无法达到设定阈值时,认为第二声音信号不满足第二预设条件。其中,设定阈值可与实际播放效果关联。
在该步骤中,第二位置用于表示第二声音信号的发出位置。
因此,在会场中第二位置拾取的第二声音信号不满足第二预设条件时,在第二位置周围确定包括第二预设范围。从而,控制第二预设范围内的第二声音拾取装置同第一声音拾取装置一起,拾取第二声音信号,进而改善第二位置拾取到的第二声音信号,使其满足第二预设条件。进一步地,在第二声音信号满足第二预设条件的情况下,可改善因拾取声音信号不好,导致播放效果不佳的现象,这样,使得讲述者讲述的内容能够清晰地传递至会场的每一位听众。
在本申请的实施例中,至少两个声音播放装置和至少两个声音拾取装置处于目标网络中,从而在目标网络中,可针对全部的声音播放装置和声音拾取装置,进行统一管理控制。一方面,在第一声音播放装置作为公共设备播放第一声音信号的情况下,若检测出会场中的第一位置处,接收到的第一声音信号不满足第一预设条件,如使得第一位置处的用户听不清楚第一声音信号,从而可在第一位置处,确定第一预设范围,再在第一预设范围中,调用其它的第二声 音播放装置,同第一声音播放装置同步播放第一声音信号,进而使得第一位置处的用户可以听清楚第一声音信号。另一方面,在第一声音拾取装置拾取第二声音信号的情况下,若拾取的第二声音信号不满足第二预设条件,如播放后不能传递至每一位听众,则可以在拾取第二声音信号的第二位置处,确定第二预设范围,再在第二预设范围中,调用其它的第二声音拾取装置,同第一声音拾取装置同步拾取第二声音信号,进而使得第二声音信号播放后能够传递至每一位听众。可见,本实施例从上述声音播放和声音拾取两个方面,确保会场中的讲述者讲述的内容能够清晰地传递至会场的每一位听众。
在本申请另一个实施例的控制方法的流程中,第一预设条件包括第一预设声音等级。相应地,S2可以包括步骤A1、A2和A3:
A1:根据第一位置接收到的第一声音信号的第一声音等级,以及第一预设声音等级,确定第一声音补偿量。
在本实施例中,可以依据声音的音质、强度等的不同程度,将声音划分为多个等级。
可选地,第一预设声音等级用于表示可达到一定音质,以及可达到一定强度的声音等级,该声音等级下的声音信号,可以确保用户能够听清楚。更进一步地,该声音等级下的声音信号,可以确保用户能够听到优质声音。
对应地,可以获取第一位置处的第一声音信号的第一声音等级,以确定第一位置处的第一声音信号的音质和强度。
在该步骤中,可基于第一声音等级和第一预设声音等级,分析欠缺量级,确定第一声音补偿量。
可选地,第一声音补偿量等于第一声音等级与第一预设声音等级之间的相差量级。
A2:根据第一声音补偿量和第一位置周边的第二声音播放装置的密集程度中的至少一项,确定以第一位置为圆心,第一距离为半径的第一预设范围。
在该步骤中,通过欠缺量级圈定补偿范围半径,其中,补偿范围半径即第一距离。
其中,若欠缺量级越大,则第一距离越大,可以使得更多的用户的设备能够处于第一预设范围内;反之,若欠缺量级越小,则第一距离越小。
另外,第一距离还关联于第一位置周围的设备分布情况。例如,第一位置周围的设备分布密集,则第一距离相对较小;反之,第一位置周围的设备分布分散,则第一距离相对较大,以使得更多的用户的设备能够处于第一预设范围内。
可选地,第一预设声音等级的设定标准可包括:优质声音的等级和基础声音的等级中的至少一种。示例性地,在确定第一预设范围时,尽可能使第一位置处的第一声音信号达到优质声音的等级;而结合第一位置的实际情况,如角落,以及第一位置周围的声音播放装置的分布情况,如分布稀少,则至少使第一位置处的第一声音信号达到基础声音的等级。
参见图2,可参考地,目标听众所在的位置为第一位置,第一位置所在的圆形为确定的第一预设范围。其中,图2中所示的其他听众为会场内除协助听众和目标听众以外的其他听众。
A3:根据第一预设范围的各个半径上的第二声音播放装置的数量,以及各个第二声音播放装置与第一位置之间的距离,确定各个第二声音播放装置播放第一声音信号的播放参数。
其中,播放参数至少包括播放音量和播放方向。
在该步骤中,第一预设范围内的个人设备的扬声器功能被调用起。进一步地,各个第二声音播放装置,根据与第一位置之间的位置距离,提供不同音量和发音方向的补偿音。这样,结合补偿音与会场公共音响的综合声音,进行及时重复音质评比,以保证第一位置的目标听众处于优质音效范围。
具体地,单条半径上补偿的设备越少,远处的设备就会强烈些补偿,保持听众耳朵四周声音均衡。
进一步地,实时根据音质分析模块分析结果,实时确定音质补偿方案。其中的音质补偿方案至少包括补偿半径和补偿半径上的各个第二声音播放装置的播放参数。
在本实施例中,针对声音播放的改善场景,提供了针对音质盲点具体地音质补偿方案,以使得会场中的音质盲点的目标听众可处于优质音效范围,从而确保会场中的讲述者讲述的内容能够清晰地传递至会场的每一位听众。
在本申请另一个实施例的控制方法的流程中,第二预设条件包括第二预设声音等级。相应地,S3可以包括步骤B1和B2:
B1:根据第一声音拾取装置拾取到的第二声音信号的第二声音等级,以及第二预设声音等级,确定第二声音补偿量。
可选地,第二预设声音等级用于表示可达到一定音质,以及可达到一定强度的声音等级,该声音等级下的声音信号,其播放后可确保会场内的用户能够听清楚。更进一步地,该声音等级下的声音信号,其播放后可确保用户能够听到优质声音。
在该步骤中,可以基于第二声音等级和第二预设声音等级,分析欠缺量级,确定第二声音补偿量。
可选地,第二声音补偿量是等于第二声音等级与第二预设声音等级之间的相差量级。
B2:根据第二声音补偿量和第二位置周边的第二声音拾取装置的密集程度至的至少一项,确定以第二位置为圆心,第二距离为半径的第二预设范围。
在该步骤中,通过欠缺量级圈定补偿范围半径,即第二距离。
其中,欠缺量级越大,则第二距离越大,以使得更多的用户的设备能够处于第二预设范围内;反之,欠缺量级越小,则第二距离越小。
另外,第二距离还关联于第二位置周围的设备分布情况。例如,第二位置周围的设备分布密集,则第二距离相对较小;反之,第二位置周围的设备分布分散,则第二距离相对较大,以使得更多的用户的设备能够处于第二预设范围内。
可选地,第二预设声音等级的设定标准可以包括:优质声音的等级和基础声音的等级。示例性地,在确定第二预设范围时,尽可能使第二声音信号播放后可达到优质声音的等级;而结合第二位置的实际情况,如第二位置周围的声 音拾取装置的分布稀少,则至少使第二声音信号播放后可达到基础声音的等级。
通常,讲述者在舞台上讲述时,由公共的话筒进行声音拾取,一般可以满足第二预设声音等级。因此,本实施例更适用于互动环节,听众发言的场景。
在互动环节,听众发言的场景中,如果是单点音源,即一个听众发言,根据补偿对象(第一声音拾取装置)的音源缺乏量级进行补偿集群(第二声音拾取装置)确定,并动态变更补偿参数,利用补偿对象周边手机或话筒进行采音,形成较好收音效果。进一步地,根据周边场景变化,进行动态调节。
如果存在多点音源段时间交替发言(如对话),则启动多点音源质量分别评估,并制定补偿方案,并根据具体发言者进行针对性补偿。如果是单点发言,就不用做交替方案。
参见图3,具体地,在两个发言者交替发言场景,进行对点音源质量评估,确定了分别缺乏的量级。由于两者位置周边的其他听众密集程度不一样,故进行位置距离半径和补偿参数的差异化设置(如图中左侧的发言者周边用户相对较少则扩大补偿半径)。具体发言时候,根据发言对象的切换,进行动态补偿协助听众圈切换。其中,图3中所示的其他听众为会场内除协助听众以外的其他听众。
其中,补偿参数包括第二距离。
在本实施例中,针对声音拾取的改善场景,提供了针对拾取效果不佳的具体音质补偿方案。其中,具备更加智能的采音倾向,可根据用户发声位置进行局部设备收音启动,并根据声音强度和质量进行周边设备补偿发言者,以使得互动环节中,发言内容能够清晰地传递至会场的每一位听众。进一步地,本实施例具备更加智能的人声分析能力,多人交替发言时候,可根据音量和音色自动区分音源,并启动周边设备进行动态收音。而相比于现有技术中,通过在听众席中间传递话筒的方式来收音,本实施例的收音方法更加便捷、智能化,从而达到提高听众的发言积极性、减少成本、提高发言效率等效果,使得会场更加高档化。
在本申请另一个实施例的控制方法的流程中,在第一声音播放装置播放第一声音信号的情况下,还包括步骤C1、C2和C3:
C1:确定第一声音信号的第三声音等级。
可选地,第三声音等级为第一声音信号可达到的声音等级,该声音等级具有一定的音质、强度等。
C2:获取第一声音播放装置的第三位置。
C3:根据第三声音等级和第三位置,确定目标网络中的各个位置接收到的第一声音信号是否满足第一预设条件。
第一声音播放装置的第三位置可展示出会场内的音响分布,从而结合拾取的第一声音信号本身的第三声音等级,可进行全场音效结构质量分析。
进一步地,目标网络中的各个位置包括加入集群的节点,即会场内的个人设备的位置。
例如,第一声音播放装置播放的第一声音信号,音质较好、强度也较高,而会场内公共音响均匀分布在各个角落里,从而会场内的各个位置均可听清楚。又如,第一声音播放装置播放的第一声音信号,强度比较低,即使会场内公共音响均匀分布在各个角落里,会场内还是会出现音质盲点。又如,会场内公共音响不均匀,而第一声音播放装置播放的第一声音信号,强度又比较低,会场必然会出现音质盲点。
在本实施例中,首先结合听众的手机资源建立局域音响集群,然后结合听众节点位置和全场音效结构质量进行综合分析,初步进行声音接收点模型建立,从而快速、准确地识别出音质盲点,进而充分利用现有资源,动态地进行放音质量评估与补偿方案调整,对音质盲点进行声音补偿,以解决边角位置等音质盲点音效缺乏的问题。
在本申请另一个实施例的控制方法的流程中,S3之前,还包括步骤D1:
D1:在第二位置处,人声声音信号的分贝值大于环境声音信号的分贝值的情况下,控制第一声音拾取装置拾取第二声音信号。
其中,第二声音信号包括环境声音信号和人声声音信号。
在会场中,若检测到某一位置处,人声声音信号的分贝值大于环境声音信号的分贝值,则认为该位置处,有人正在发言,从而控制该位置处的第一声音拾取装置拾取第二声音信号。
一种方式为:根据第二位置,识别对应的第一声音拾取装置,控制第一声音拾取装置的麦克风功能开启;另一种方式为:当人声声音信号达到一定的预设分贝值时,自动激活对应位置的第一声音拾取装置的麦克风功能。
可选地,第二位置的数量包括多个。例如,在发言场景中,多个位置处的听众进行交替发言。
进一步地,对于每个第二位置,其对应第一声音拾取装置拾取该位置处的第二声音信号的情况下,针对该位置处的第二声音信号,进行声音分析和补偿。当多个第二位置处交替出现发言者时,则交替进行声音分析和补偿。同时,在进行声音补偿的情况下,根据实时情况,调整补偿方案。
其中,对于不同位置处对应不同的第二声音信号,可通过音色等特征来区分不同的第二声音信号。
因此,本实施例还可结合不同第二位置处的第二声音信号的音色等特征信息,针对不同位置的不同发言者,进行声音分析和补偿。
在本实施例中,提供了一种检测第二位置的方法。当提问环节时候,将进行基础会场音源分析,确定发言者位置、音源质量、以及音源数量,以快速、准确地识别出发言位置,从而对发言位置进行声音分析和补偿,进而结合听众的手机资源建立局域话筒集群,充分利用现有资源,动态的进行采音质量评估与补偿方案调整,降低了话筒设备与传递的人力成本。
综上,在本申请的实施例中,通过手机局域集群网,智能分析并采集音源和传声,针对会场传声与采音进行补偿操作,提升了会场听众听音质量以及发言场景提问互动效率与积极性。其中,在本申请的实施例中,其一,会场音响和听众的手机都可以作为音响,可根据声音在会场中分布情况,进行均匀化调整,提供了更加均匀的音响集群;其二,讲述者的话筒与听众的手机处于同一个局域网络环境下,设备都可以进行音源收录,形成数量众多并分布均匀的话 筒集群,提供了更加便捷的连接方式和均匀的话筒集群。另外,在本申请的实施例中,第一方面,本申请的实施例智能、及时的质量评估与资源调动方案设计,将整个会场与实际设备、人员周边环境紧密连接,适用范围与能力极强;第二方面,总体上解决了传递话筒的麻烦、让全场全员都可以高质量的参与到互动中来。
在本申请的实施例提供的控制方法的基础上,此种集群设备资源智能调动方案,可利用到城市环境光源设计,根据实际质量与需求进行动态调整光源调用对象以及对应参数。
需要说明的是,本申请实施例提供的控制方法,执行主体可以为控制装置,或者该控制装置中的用于执行控制方法的控制模块。本申请实施例中以控制装置执行控制方法为例,说明本申请实施例提供的控制方法的控制装置。
图4示出了本申请提供的另一个实施例的控制装置的框图,包括:
识别模块10,用于识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;
播放补偿模块20,用于在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放第一声音信号;第一预设范围与第一位置相关联;
拾音补偿模块30,用于在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取第二声音信号;第二预设范围与第二位置相关联。
这样,在本申请的实施例中,至少两个声音播放装置和至少两个声音拾取装置处于目标网络中,从而在目标网络中,可针对全部的声音播放装置和声音拾取装置,进行统一管理控制。一方面,在第一声音播放装置作为公共设备播放第一声音信号的情况下,若检测出会场中的第一位置处,接收到的第一声音信号不满足第一预设条件,如使得第一位置处的用户听不清楚第一声音信号,从而可在第一位置处,确定第一预设范围,再在第一预设范围中,调用其它的第二声音播放装置,同第一声音播放装置同步播放第一声音信号,进而使得第 一位置处的用户可以听清楚第一声音信号。另一方面,在第一声音拾取装置拾取第二声音信号的情况下,若拾取的第二声音信号不满足第二预设条件,如播放后不能传递至每一位听众,从而可在拾取第二声音信号的第二位置处,确定第二预设范围,再在第二预设范围中,调用其它的第二声音拾取装置,同第一声音拾取装置同步拾取第二声音信号,进而使得第二声音信号播放后能够传递至每一位听众。可见,本实施例从上述声音播放和声音拾取两个方面,确保会场中的讲述者讲述的内容能够清晰地传递至会场的每一位听众。
可选地,第一预设条件包括第一预设声音等级;播放补偿模块20,包括:
第一确定单元,用于根据第一位置接收到的第一声音信号的第一声音等级,以及第一预设声音等级,确定第一声音补偿量;
第二确定单元,用于根据第一声音补偿量和第一位置周边的第二声音播放装置的密集程度中的至少一项,确定以第一位置为圆心,第一距离为半径的第一预设范围;
第三确定单元,用于根据第一预设范围的各个半径上的第二声音播放装置的数量,以及各个第二声音播放装置与第一位置之间的距离,确定各个第二声音播放装置播放第一声音信号的播放参数;播放参数至少包括播放音量和播放方向。
可选地,第二预设条件包括第二预设声音等级;拾音补偿模块30,包括:
第四确定单元,用于根据第一声音拾取装置拾取到的第二声音信号的第二声音等级,以及第二预设声音等级,确定第二声音补偿量;
第五确定单元,用于根据第二声音补偿量和第二位置周边的第二声音拾取装置的密集程度至的至少一项,确定以第二位置为圆心,第二距离为半径的第二预设范围。
可选地,装置,还包括:
第六确定模块,用于确定第一声音信号的第三声音等级;
获取模块,用于获取第一声音播放装置的第三位置;
第七确定模块,用于根据第三声音等级和第三位置,确定目标网络中的各 个位置接收到的第一声音信号是否满足第一预设条件。
可选地,装置,还包括:
拾取模块,用于在第二位置处,人声声音信号的分贝值大于环境声音信号的分贝值的情况下,控制第一声音拾取装置拾取第二声音信号;第二声音信号包括环境声音信号和人声声音信号。
本申请实施例中的控制装置可以是装置,也可以是终端中的部件、集成电路、或芯片。该装置可以是移动电子设备,也可以为非移动电子设备。示例性的,移动电子设备可以为手机、平板电脑、笔记本电脑、掌上电脑、车载电子设备、可穿戴设备、超级移动个人计算机(ultra-mobile personal computer,UMPC)、上网本或者个人数字助理(personal digital assistant,PDA)等,非移动电子设备可以为服务器、网络附属存储器(Network Attached Storage,NAS)、个人计算机(personal computer,PC)、电视机(television,TV)、柜员机或者自助机等,本申请实施例不作具体限定。
本申请实施例中的控制装置可以为具有操作系统的装置。该操作系统可以为安卓(Android)操作系统,可以为ios操作系统,还可以为其他可能的操作系统,本申请实施例不作具体限定。
本申请实施例提供的控制装置能够实现上述方法实施例实现的各个过程,为避免重复,这里不再赘述。
可选地,如图5所示,本申请实施例还提供一种电子设备100,包括处理器101,存储器102,存储在存储器102上并可在所述处理器101上运行的程序或指令,该程序或指令被处理器101执行时实现上述任一种控制方法实施例的各个过程,且能达到相同的技术效果,为避免重复,这里不再赘述。
需要说明的是,本申请实施例中的电子设备包括上述所述的移动电子设备和非移动电子设备。
图6为实现本申请实施例的一种电子设备的硬件结构示意图。
该电子设备1000包括但不限于:射频单元1001、网络模块1002、音频输出单元1003、输入单元1004、传感器1005、显示单元1006、用户输入单元 1007、接口单元1008、存储器1009、处理器1010等部件。
本领域技术人员可以理解,电子设备1000还可以包括给各个部件供电的电源(比如电池),电源可以通过电源管理系统与处理器1010逻辑相连,从而通过电源管理系统实现管理充电、放电、以及功耗管理等功能。图6中示出的电子设备结构并不构成对电子设备的限定,电子设备可以包括比图示更多或更少的部件,或者组合某些部件,或者不同的部件布置,在此不再赘述。
其中,处理器1010,用于识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放第一声音信号;第一预设范围与第一位置相关联;在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取第二声音信号;第二预设范围与第二位置相关联。
这样,在本申请的实施例中,至少两个声音播放装置和至少两个声音拾取装置处于目标网络中,从而在目标网络中,可针对全部的声音播放装置和声音拾取装置,进行统一管理控制。一方面,在第一声音播放装置作为公共设备播放第一声音信号的情况下,若检测出会场中的第一位置处,接收到的第一声音信号不满足第一预设条件,如使得第一位置处的用户听不清楚第一声音信号,从而可在第一位置处,确定第一预设范围,再在第一预设范围中,调用其它的第二声音播放装置,同第一声音播放装置同步播放第一声音信号,进而使得第一位置处的用户可以听清楚第一声音信号。另一方面,在第一声音拾取装置拾取第二声音信号的情况下,若拾取的第二声音信号不满足第二预设条件,如播放后不能传递至每一位听众,从而可在拾取第二声音信号的第二位置处,确定第二预设范围,再在第二预设范围中,调用其它的第二声音拾取装置,同第一声音拾取装置同步拾取第二声音信号,进而使得第二声音信号播放后能够传递至每一位听众。可见,本实施例从上述声音播放和声音拾取两个方面,确保会场中的讲述者讲述的内容能够清晰地传递至会场的每一位听众。
可选地,第一预设条件包括第一预设声音等级;处理器1010,还用于根据第一位置接收到的第一声音信号的第一声音等级,以及第一预设声音等级,确定第一声音补偿量;根据第一声音补偿量和第一位置周边的第二声音播放装置的密集程度中的至少一项,确定以第一位置为圆心,第一距离为半径的第一预设范围;根据第一预设范围的各个半径上的第二声音播放装置的数量,以及各个第二声音播放装置与第一位置之间的距离,确定各个第二声音播放装置播放第一声音信号的播放参数;播放参数至少包括播放音量和播放方向。
可选地,第二预设条件包括第二预设声音等级;处理器1010,还用于根据第一声音拾取装置拾取到的第二声音信号的第二声音等级,以及第二预设声音等级,确定第二声音补偿量;根据第二声音补偿量和第二位置周边的第二声音拾取装置的密集程度至的至少一项,确定以第二位置为圆心,第二距离为半径的第二预设范围。
可选地,处理器1010,还用于确定第一声音信号的第三声音等级;获取第一声音播放装置的第三位置;根据第三声音等级和第三位置,确定目标网络中的各个位置接收到的第一声音信号是否满足第一预设条件。
可选地,处理器1010,还用于在第二位置处,人声声音信号的分贝值大于环境声音信号的分贝值的情况下,控制第一声音拾取装置拾取第二声音信号;第二声音信号包括环境声音信号和人声声音信号。
综上,在本申请的实施例中,通过手机局域集群网,智能分析并采集音源和传声,针对会场传声与采音进行补偿操作,提升了会场听众听音质量以及发言场景提问互动效率与积极性。其中,在本申请的实施例中,其一,会场音响和听众的手机都可以作为音响,可根据声音在会场中分布情况,进行均匀化调整,提供了更加均匀的音响集群;其二,讲述者的话筒与听众的手机处于同一个局域网络环境下,设备都可以进行音源收录,形成数量众多并分布均匀的话筒集群,提供了更加便捷的连接方式和均匀的话筒集群。另外,在本申请的实施例中,第一方面,本申请的实施例智能、及时的质量评估与资源调动方案设计,将整个会场与实际设备、人员周边环境紧密连接,适用范围与能力极强; 第二方面,总体上解决了传递话筒的麻烦、让全场全员都可以高质量的参与到互动中来。
应理解的是,本申请实施例中,输入单元1004可以包括图形处理器(Graphics Processing Unit,GPU)10041和麦克风10042,图形处理器10041对在视频捕获模式或图像捕获模式中由图像捕获装置(如摄像头)获得的静态图片或视频的图像数据进行处理。显示单元1006可包括显示面板10061,可以采用液晶显示器、有机发光二极管等形式来配置显示面板10061。用户输入单元1007包括触控面板10071以及其他输入设备10072。触控面板10071,也称为触摸屏。触控面板10071可包括触摸检测装置和触摸控制器两个部分。其他输入设备10072可以包括但不限于物理键盘、功能键(比如音量控制按键、开关按键等)、轨迹球、鼠标、操作杆,在此不再赘述。
存储器1009可用于存储软件程序以及各种数据,包括但不限于应用程序和操作系统。处理器1010可集成应用处理器和调制解调处理器,其中,应用处理器主要处理操作系统、用户界面和应用程序等,调制解调处理器主要处理无线通信。可以理解的是,上述调制解调处理器也可以不集成到处理器1010中。本申请实施例还提供一种可读存储介质,可读存储介质上存储有程序或指令,该程序或指令被处理器执行时实现上述任一种控制方法实施例的各个过程,且能达到相同的技术效果,为避免重复,这里不再赘述。
其中,处理器为上述实施例中的电子设备中的处理器。可读存储介质的示例包括计算机可读存储介质,如计算机只读存储器(Read-Only Memory,ROM)、随机存取存储器(Random Access Memory,RAM)、磁碟或者光盘等的非暂态计算机可读存储介质。
本申请实施例另提供了一种芯片,芯片包括处理器和通信接口,通信接口和处理器耦合,处理器用于运行程序或指令,实现上述任一种控制方法实施例的各个过程,且能达到相同的技术效果,为避免重复,这里不再赘述。
应理解,本申请实施例提到的芯片还可以称为系统级芯片、系统芯片、芯片系统或片上系统芯片等。
需要说明的是,在本文中,术语“包括”、“包含”或者其任何其他变体意在涵盖非排他性的包含,从而使得包括一系列要素的过程、方法、物品或者装置不仅包括那些要素,而且还包括没有明确列出的其他要素,或者是还包括为这种过程、方法、物品或者装置所固有的要素。在没有更多限制的情况下,由语句“包括一个……”限定的要素,并不排除在包括该要素的过程、方法、物品或者装置中还存在另外的相同要素。此外,需要指出的是,本申请实施方式中的方法和装置的范围不限按示出或讨论的顺序来执行功能,还可包括根据所涉及的功能按基本同时的方式或按相反的顺序来执行功能,例如,可以按不同于所描述的次序来执行所描述的方法,并且还可以添加、省去、或组合各种步骤。另外,参照某些示例所描述的特征可在其他示例中被组合。
通过以上的实施方式的描述,本领域的技术人员可以清楚地了解到上述实施例方法可借助软件加必需的通用硬件平台的方式来实现,当然也可以通过硬件,但很多情况下前者是更佳的实施方式。基于这样的理解,本申请的技术方案本质上或者说对现有技术做出贡献的部分可以以软件产品的形式体现出来,该计算机软件产品存储在一个存储介质(如ROM/RAM、磁碟、光盘)中,包括若干指令用以使得一台终端(可以是手机,计算机,服务器,空调器,或者网络设备等)执行本申请各个实施例所述的方法。
上面结合附图对本申请的实施例进行了描述,但是本申请并不局限于上述的具体实施方式,上述的具体实施方式仅仅是示意性的,而不是限制性的,本领域的普通技术人员在本申请的启示下,在不脱离本申请宗旨和权利要求所保护的范围情况下,还可做出很多形式,均属于本申请的保护之内。

Claims (15)

  1. 一种控制方法,包括:
    识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;
    在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的所述第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放所述第一声音信号;所述第一预设范围与所述第一位置相关联;
    在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若所述第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取所述第二声音信号;所述第二预设范围与所述第二位置相关联。
  2. 根据权利要求1所述的方法,其中,所述第一预设条件包括第一预设声音等级;所述控制第一预设范围内的第二声音播放装置播放所述第一声音信号,包括:
    根据所述第一位置接收到的所述第一声音信号的第一声音等级,以及所述第一预设声音等级,确定第一声音补偿量;
    根据所述第一声音补偿量和所述第一位置周边的所述第二声音播放装置的密集程度中的至少一项,确定以所述第一位置为圆心,第一距离为半径的第一预设范围;
    根据所述第一预设范围的各个半径上的第二声音播放装置的数量,以及各个所述第二声音播放装置与所述第一位置之间的距离,确定各个所述第二声音播放装置播放所述第一声音信号的播放参数;所述播放参数至少包括播放音量和播放方向。
  3. 根据权利要求1所述的方法,其中,所述第二预设条件包括第二预设声音等级;所述控制第二预设范围内的第二声音拾取装置拾取所述第二声音信号,包括:
    根据所述第一声音拾取装置拾取到的所述第二声音信号的第二声音等级,以及所述第二预设声音等级,确定第二声音补偿量;
    根据所述第二声音补偿量和所述第二位置周边的所述第二声音拾取装置 的密集程度至的至少一项,确定以所述第二位置为圆心,第二距离为半径的第二预设范围。
  4. 根据权利要求1所述的方法,其中,在第一声音播放装置播放第一声音信号的情况下,所述方法还包括:
    确定所述第一声音信号的第三声音等级;
    获取所述第一声音播放装置的第三位置;
    根据所述第三声音等级和所述第三位置,确定所述目标网络中的各个位置接收到的所述第一声音信号是否满足所述第一预设条件。
  5. 根据权利要求1所述的方法,其中,在所述在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若所述第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取所述第二声音信号之前,所述方法还包括:
    在所述第二位置处,人声声音信号的分贝值大于环境声音信号的分贝值的情况下,控制所述第一声音拾取装置拾取所述第二声音信号;所述第二声音信号包括所述环境声音信号和所述人声声音信号。
  6. 一种控制装置,包括:
    识别模块,用于识别处于目标网络中的至少两个声音播放装置和至少两个声音拾取装置;
    播放补偿模块,用于在第一声音播放装置播放第一声音信号的情况下,若第一位置接收到的所述第一声音信号不满足第一预设条件,则控制第一预设范围内的第二声音播放装置播放所述第一声音信号;所述第一预设范围与所述第一位置相关联;
    拾音补偿模块,用于在第二位置对应的第一声音拾取装置拾取第二声音信号的情况下,若所述第二声音信号不满足第二预设条件,则控制第二预设范围内的第二声音拾取装置拾取所述第二声音信号;所述第二预设范围与所述第二位置相关联。
  7. 根据权利要求6所述的装置,其中,所述第一预设条件包括第一预设声 音等级;所述播放补偿模块包括:
    第一确定单元,用于根据所述第一位置接收到的所述第一声音信号的第一声音等级,以及所述第一预设声音等级,确定第一声音补偿量;
    第二确定单元,用于根据所述第一声音补偿量和所述第一位置周边的所述第二声音播放装置的密集程度中的至少一项,确定以所述第一位置为圆心,第一距离为半径的第一预设范围;
    第三确定单元,用于根据所述第一预设范围的各个半径上的第二声音播放装置的数量,以及各个所述第二声音播放装置与所述第一位置之间的距离,确定各个所述第二声音播放装置播放所述第一声音信号的播放参数;所述播放参数至少包括播放音量和播放方向。
  8. 根据权利要求6所述的装置,其中,所述第二预设条件包括第二预设声音等级;所述拾音补偿模块包括:
    第四确定单元,用于根据所述第一声音拾取装置拾取到的所述第二声音信号的第二声音等级,以及所述第二预设声音等级,确定第二声音补偿量;
    第五确定单元,用于根据所述第二声音补偿量和所述第二位置周边的所述第二声音拾取装置的密集程度至的至少一项,确定以所述第二位置为圆心,第二距离为半径的第二预设范围。
  9. 根据权利要求6所述的装置,还包括:
    第六确定模块,用于确定所述第一声音信号的第三声音等级;
    获取模块,用于获取所述第一声音播放装置的第三位置;
    第七确定模块,用于根据所述第三声音等级和所述第三位置,确定所述目标网络中的各个位置接收到的所述第一声音信号是否满足所述第一预设条件。
  10. 根据权利要求6所述的装置,还包括:
    拾取模块,用于在所述第二位置处,人声声音信号的分贝值大于环境声音信号的分贝值的情况下,控制所述第一声音拾取装置拾取所述第二声音信号;所述第二声音信号包括所述环境声音信号和所述人声声音信号。
  11. 一种电子设备,包括处理器,存储器及存储在所述存储器上并可在所述处理器上运行的程序或指令,所述程序或指令被所述处理器执行时实现如权利要求1-5任一项所述的控制方法的步骤。
  12. 一种电子设备,被配置用于实现如权利要求1-5任一项所述的控制方法。
  13. 一种可读存储介质,所述可读存储介质上存储有程序或指令,所述程序或指令被处理器执行时实现如权利要求1至5中任一项所述的控制方法。
  14. 一种芯片,包括处理器和通信接口,通信接口和处理器耦合,处理器用于运行程序或指令,实现如权利要求1-5任一项所述的控制方法。
  15. 一种计算机程序产品,其特征在于,所述计算机程序产品中的指令由电子设备的处理器执行时,使得所述电子设备执行如权利要求1-5任意一项所述的控制方法。
PCT/CN2022/074425 2021-01-28 2022-01-27 控制方法、装置和电子设备 WO2022161446A1 (zh)

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