CN218162505U

CN218162505U - Call system and host device

Info

Publication number: CN218162505U
Application number: CN202220384362.5U
Authority: CN
Inventors: 刘德志; 周强
Original assignee: Shenzhen Hollyland Technology Co Ltd
Current assignee: Shenzhen Hollyland Technology Co Ltd
Priority date: 2022-02-24
Filing date: 2022-02-24
Publication date: 2022-12-27
Anticipated expiration: 2032-02-24

Abstract

The embodiment of the specification provides a communication system and host equipment. The communication system comprises a host device, a cloud service end and a plurality of client sides, wherein the host device is used for receiving first voice signals sent by the plurality of terminal devices, receiving coded second voice signals sent by the plurality of client sides and forwarded by the cloud service end, decoding the coded second voice signals, mixing the received first voice signals and the decoded second voice signals, respectively generating a mixed voice signal corresponding to one of the terminal devices, sending the mixed voice signal to the terminal device, generating a mixed voice signal corresponding to one of the client sides, and forwarding the mixed voice signal to the client sides through the cloud service end. The communication system can realize that users which are not in the coverage range of the internal communication system are accessed into the communication system to communicate.

Description

Call system and host device

Technical Field

The present specification relates to the field of communications technologies, and in particular, to a call system and a host device.

Background

In scenes such as movie shooting, stage art performance, large events and the like, an intercom system is generally required to be used for realizing communication among field workers so as to complete organization and scheduling of work. Generally, an intercom system includes a host device and a plurality of terminal devices (e.g., an intercom, or a belt pack for conversation, etc.) capable of being in communication connection with the host device, where the terminal devices can send voice signals of users to the host device, and the host device can perform mixing processing on the received voice signals, that is, for each terminal device, after mixing the voice signals sent by other terminal devices except the terminal device, the mixed voice signals are forwarded to the terminal device. Because the host device and the terminal device are usually connected wirelessly or by wire, and because the wireless communication distance is limited and the deployment distance of the wire connection is also limited, the intercom system can only be applied in a small range, and users outside the communication coverage range of the intercom system can not be accessed into the intercom system to communicate with the terminal device in the intercom system. In addition, if the intercom system is a wireless intercom system, the number of access terminal devices in the wireless intercom system is also limited due to the limited wireless communication channel resources that are usually available, and the wireless intercom system cannot be applied to a scenario that requires a large number of users to participate in a call.

SUMMERY OF THE UTILITY MODEL

Based on this, this specification implementation provides calling system and host equipment.

According to a first aspect of embodiments of the present specification, a call system is provided, which includes a host device, a cloud server, and a plurality of clients, where the host device is in communication connection with a plurality of terminal devices; the host equipment and the client establish a connection channel with the cloud server through the Internet respectively;

the terminal equipment is used for acquiring a first voice signal and sending the first voice signal to the host equipment;

the client is used for acquiring a second voice signal, encoding the second voice signal and then sending the encoded second voice signal to the cloud server;

the cloud server is used for receiving the encoded second voice signal and forwarding the encoded second voice signal to the host equipment;

the host equipment is used for receiving the first voice signal and the encoded second voice signal, decoding the encoded second voice signal, mixing the received first voice signal and the decoded second voice signal, generating a mixed voice signal corresponding to one of the terminal equipment, sending the mixed voice signal to the terminal equipment, generating a mixed voice signal corresponding to one of the client sides, and forwarding the mixed voice signal to the client side through the cloud server side.

According to a second aspect of embodiments herein, there is provided a host device, the host device being communicatively connected to a plurality of terminal devices; the host equipment and the plurality of clients respectively establish a connecting channel with the cloud server through the Internet; the host device comprises a communication module, a coding and decoding module and a sound mixing processing module,

the communication module is used for receiving a first voice signal sent by the terminal device, receiving a second voice signal which is forwarded by the cloud server and is sent by the client after being encoded, transmitting the first voice signal to the audio mixing processing module, transmitting the second voice signal after being encoded to the encoding and decoding module, receiving an audio mixing signal which is transmitted by the audio mixing processing module and corresponds to one of the terminal devices, sending the audio mixing signal to the terminal device, receiving an audio mixing signal which is transmitted by the audio mixing processing module and corresponds to one of the client devices, and sending the audio mixing signal which is transmitted by the audio mixing processing module and corresponds to the one of the client devices to the client;

the encoding and decoding module is used for decoding the received encoded second voice signal and transmitting the decoded second voice signal to the audio mixing processing module, receiving the audio mixing signal which is transmitted by the audio mixing processing module and corresponds to one of the clients, and encoding the audio mixing signal so as to send the encoded audio mixing signal to the cloud server;

the audio mixing processing module is used for performing audio mixing processing on the first voice signal and a second voice signal obtained by decoding, and generating an audio mixing signal corresponding to one of the terminal devices and an audio mixing signal corresponding to one of the clients.

According to a third aspect of the embodiments of the present specification, there is provided a call system, including a host device, a plurality of terminal devices, a cloud server, and a plurality of clients, where the host device and the terminal devices are in the same wireless access network; the host equipment and the client establish a connection channel with the cloud server through the Internet respectively;

By applying the scheme of the embodiment of the specification, it is considered that although the current intercom system has the advantages of flexible deployment, high conversation quality and the like, the intercom system can only communicate in a small range, the coverage range is limited, and the number of the accessed terminal devices is also limited. Although the internet voice-based network communication has poor communication quality, the coverage range is wide, and wide-area deployment can be realized. Therefore, the advantages of the host equipment and the cloud service end can be combined to construct a call system, the call system comprises the host equipment, the cloud service end and a plurality of clients, the host equipment is in communication connection with a plurality of terminal equipment, and the cloud service end is in communication connection with the host equipment and the clients respectively based on the internet; the host equipment is used for receiving first voice signals sent by a plurality of terminal equipment and encoded second voice signals which are forwarded by a cloud server and sent by a plurality of clients, decoding the second voice signals, mixing the received first voice signals and the second voice signals obtained through decoding, respectively generating a mixed voice signal corresponding to one of the plurality of terminal equipment, sending the mixed voice signal corresponding to each terminal equipment to the terminal equipment, and generating a mixed voice signal corresponding to one of the plurality of clients; and forwarding the audio mixing signal corresponding to the client through the cloud server. The communication system has high-quality communication performance, can be flexibly deployed, has large-scale coverage capability, and can realize that users which are not in the coverage range of the internal communication system are accessed into the internal communication system for communication.

It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory only and are not restrictive of the specification.

Drawings

The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the present specification and together with the description, serve to explain the principles of the specification.

Fig. 1 (a) is a schematic structural diagram of a wireless intercom system according to an embodiment of the present disclosure.

Fig. 1 (b) is a schematic structural diagram of a wireless intercom system according to an embodiment of the present disclosure.

Fig. 2-6 are schematic structural diagrams of a wireless intercom system according to an embodiment of the present disclosure.

Fig. 7 is a schematic diagram of an internal structure of a host device according to an embodiment of the present disclosure.

Detailed Description

Reference will now be made in detail to the exemplary embodiments, examples of which are illustrated in the accompanying drawings. When the following description refers to the accompanying drawings, like numbers in different drawings represent the same or similar elements unless otherwise indicated. The embodiments described in the following exemplary embodiments do not represent all embodiments consistent with the present specification. Rather, they are merely examples of apparatus and methods consistent with certain aspects of the specification, as detailed in the appended claims.

The terminology used in the description herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the description. As used in this specification and the appended claims, the singular forms "a", "an", and "the" are intended to include the plural forms as well, unless the context clearly indicates otherwise. It should also be understood that the term "and/or" as used herein refers to and encompasses any and all possible combinations of one or more of the associated listed items.

It should be understood that although the terms first, second, third, etc. may be used herein to describe various information, these information should not be limited to these terms. These terms are only used to distinguish one type of information from another. For example, the first information may also be referred to as second information, and similarly, the second information may also be referred to as first information, without departing from the scope of the present specification. The word "if" as used herein may be interpreted as "at" \8230; "or" when 8230; \8230; "or" in response to a determination ", depending on the context.

In scenes such as movie shooting and large-scale activities, an intercom system is usually required to realize communication between field workers so as to complete organization and scheduling of work. Intercom systems include wired intercom systems and wireless intercom systems, and wireless intercom systems are more convenient and flexible to deploy than wired intercom systems, and are therefore widely used. As shown in fig. 1 (a), a schematic diagram of a wireless intercom system including a host device and a plurality of terminal devices wirelessly connected to the host device is shown. In some scenarios, the terminal device may be an integrated device that integrates a voice signal collecting function, a voice signal playing function, and a wireless transceiving function, for example, a handheld interphone. In some scenarios, the terminal device may also be a device having only a wireless transceiving function, and the terminal device may be connected to the voice acquisition and voice playing device, and configured to receive a voice signal acquired by the voice acquisition device, forward the voice signal to the host device, receive a mixed sound signal sent by the host device, and play the mixed sound signal to a user through the voice playing device. For example, as shown in fig. 1 (b), the terminal device may be a belt pack, and the belt pack may be connected to a headset.

The host device may be configured to implement a sound mixing function, that is, after receiving the voice signal sent by the terminal device, the host device performs sound mixing processing on the voice signal sent by the terminal device for each terminal device, and forwards the voice signal to the terminal device. In addition, the host equipment can also be configured with functions of sound effect improvement, network expansion, analog two-four-wire voice access, digital voice expansion, call grouping, human-computer interaction interface providing and the like based on actual requirements. Of course, in practical applications, the host device may be an integrated device, or may be a device obtained by connecting and combining two or more devices in a certain manner.

Because the host device and the terminal device are usually in wireless connection and the distance of wireless communication is limited, the host device and the terminal device can only be applied in a small range, and other users outside the coverage range of wireless communication cannot be accessed into the communication system to communicate with the terminal device in the communication system. For example. For example, at a certain activity site, in addition to communication between workers at the activity site, other workers at a distance from the activity site may need to be accessed to work together. Or the staff in the activity site needs to leave the activity site temporarily due to the temporary existence, but the organization scheduling of the possible activity still needs to participate, and the current wireless communication system cannot meet the requirements of the above scenarios.

In addition, the current wireless intercom system needs to support full-duplex real-time conversation, so that the full-duplex conversation between the terminal devices can be realized only by allocating an independent wireless communication channel for the voice signals between the terminal devices and allocating an independent wireless communication channel for the uplink and downlink voice signals of each terminal device, and stable and reliable conversation performance is ensured. However, the number of terminal devices in the call system is also limited due to the limited resources of the wireless communication channel, and the method is not suitable for a scenario with a large number of call users.

Based on this, the embodiment of the present application provides a call system, and it is considered that although the current wireless intercom system has the advantages of flexible deployment, high call quality, and the like, it can only communicate within a small range, the coverage area is limited, and the number of terminal devices that are accessed is also limited. The network communication based on the internet voice has poor communication quality but wide coverage range, and can realize wide-area deployment. Therefore, the advantages of the two can be combined, a communication system which has high-quality communication performance, can be flexibly deployed and has large-scale coverage capacity is constructed, users which are not in the coverage range of the wireless internal communication system can be accessed into the communication system for communication through the communication system, and the users can be accessed into the communication system at any time and any place due to wide internet coverage range, so that the communication system is very flexible and convenient.

Meanwhile, because the network call based on the internet voice can be realized without using wireless channel resources used by the wireless internal call system, the number of call users in the call system is not limited due to the limited wireless communication channel resources, namely, the call system can accommodate more users.

As shown in fig. 2, a call system 10 provided for the embodiment of the present application includes a host device 11, a cloud server 12, and a plurality of clients 13, where the host device 11 is wirelessly connected to a plurality of terminal devices 14. The host device 11 is in communication connection with the terminal device 11, and the host device 11, the client 13 and the cloud server 12 are connected through the internet.

For example, the host device 11 and the terminal device 14 may be connected by a wired connection, and certainly, because the communication range of the wired connection is limited, the host device 11 and the terminal device 14 may also communicate by short-distance communication methods such as WiFi, bluetooth, zigBee, DECT, and the like, or in order to ensure the communication quality of the two, the host device 11 and the terminal device 14 may also implement wireless communication by some self-defined private protocols, for example, the host device 11 and the terminal device 14 may operate in an independently allocated frequency band, such as a UHF frequency band, a 1.9GHz DECT frequency band, and a 2.4G/5GHz ISM frequency band, and at the same time, a digital wireless modulation mode such as FSK, BPSK, QPSK, and the like may be employed to support ACS, ADHS automatic frequency hopping, and the like, so as to ensure that the two have higher anti-interference communication quality. The voice signals between the terminal devices can be allocated with separate wireless communication channels, and the uplink and downlink voice signals of each terminal device can be allocated with separate wireless communication channels, so as to realize full-duplex communication between the terminal devices.

The cloud server 12 is in communication connection with the host device 11 and the plurality of clients 13 respectively based on an internet protocol. In order to enable other users far away from the host device 11 (for example, other users who cannot perform wired connection with the host device 11 or are not within the wireless network coverage range of the wireless intercom system formed by the terminal device 14 and the host device 11) to join the talk group formed by the terminal devices for voice talk, the host device 11 and the cloud server 12 may be in communication connection through the internet, and the cloud server 12 and the client 13 may also be in communication connection through the internet. Because the internet coverage is wide, the client 13 can be accessed to the call group formed by the terminal devices 14 at any time and any place, and realizes group call with the terminal devices 14, which is very convenient and flexible.

For example, the plurality of terminal devices 14 may collect a first voice signal of a user in an activity field and send the first voice signal to the host device 11, and the plurality of clients 13 may collect a second voice signal of a user (of course, a user in an activity field) far away from the activity field and send the second voice signal to the cloud server 12, and the second voice signal is forwarded to the host device 11 by the cloud server 12. The host device 11 may receive the first voice signal and the second voice signal, and perform sound mixing processing on the received first voice signal and the second voice signal, to generate a sound mixing signal corresponding to one of the plurality of terminal devices 14 (of course, each terminal device of the communication system may also be), and a sound mixing signal corresponding to one of the plurality of clients 13, or of course, each terminal device and each client of the communication system may also be specifically set based on actual conditions. After the mixing process is completed, the host device 11 may send the mixing signal corresponding to each terminal device 14, and forward the mixing signal corresponding to each client 13 through the cloud service 12.

For any terminal device 14, the voice signal sent by the terminal device 14 in the received voice signal may be removed, and the voice signals sent by other terminal devices 14 and the client 13 in the same group as the terminal device 14 are mixed and sent to the terminal device 14. Similarly, for any client 13, the voice signals sent by the terminal devices 14 in the same group as the client 13 and other clients 13 may be mixed and sent to the client 13. Therefore, each member in the group can hear the speech of other members, and the speech of a plurality of members is mixed into one path of speech and then transmitted, so that the transmission efficiency can be improved, and the time delay is reduced.

The terminal device 14 may be a device integrating a voice signal collecting function, a voice signal playing function, and a wireless signal transceiving function, for example, a handheld intercom device. Certainly, the terminal device 14 may also be a device having only a wireless signal transceiving function, for example, the terminal device 14 may be a belt pack, and the terminal device 14 may be connected to a device (for example, a headset) 15 having a voice signal collecting and playing function, as shown in fig. 1 (b), the headset 15 may be configured to collect a first voice signal of a user and transmit the first voice signal to the terminal device 14, and the terminal device 14 sends the mixed signal to the host device 11 for sound mixing processing, then receives the mixed signal sent by the host device 11, and forwards the mixed signal to the microphone 15 to play the mixed signal to the user.

In some scenarios, the host device 11 may be an integrated device, which, besides having the function of audio mixing processing, can integrate functions required by various communication systems, such as sound enhancement, network extension, analog two-four-wire voice access, digital voice extension, group management, human-computer interface provision, and Tally device connection, according to actual needs. Of course, in some scenarios, the host device 11 may also be a device obtained by combining two or more devices, for example, the host device 11 may be formed by two devices that are communicatively connected, and each device may implement a part of the above functions.

The cloud server 12 may be a call service provided on a cloud server or a cloud server cluster, and the call service may enable the client 13 and the terminal device 14 to perform a group call. For example, the session service may be deployed on a platform of a current mainstream cloud service provider PaSS or SaSS, such as amazon cloud, microsoft cloud, aristoloc cloud, or Tencent cloud, which is not limited in the embodiments of the present application.

The client 13 may be an APP installed on a mobile terminal, or a web page client, for example, an APP installed on a mobile terminal such as a mobile phone, a smart wearable device, a tablet, and the like. The client 13 may send the second voice signal acquired by the mobile terminal to the cloud server 12, and forward the second voice signal to the host device 11 through the cloud server 12, and meanwhile, may receive the audio mixing signal and play the audio mixing signal to the user. Of course, in some embodiments, if the terminal device 14 or the host device 11 has the function of installing APP, the client may also be installed on the terminal device 14 or the host device 11.

Because the host device 11 and the cloud service end 12 can be connected via the internet, and the cloud service end 12 and the client 13 can also be connected via the internet, a certain delay exists in the transmission of the voice signal via the internet, in order to reduce the delay as much as possible and ensure the call quality of the client 13, after the client 13 collects the second voice signal of the user, the second voice signal can be compressed and encoded first and then sent to the cloud service end 12, and correspondingly, after receiving the encoded second voice signal forwarded by the cloud service end 12, the host device 11 can decode the received encoded second voice signal first and then perform audio mixing processing on the decoded second voice signal and the decoded first voice signal. Similarly, after the mixing process is completed, the host device 11 may encode the mixing signal corresponding to each client 13 and then forward the encoded mixing signal to each client 13 through the cloud service 12. The second voice signal transmitted by the internet can be transmitted after being encoded, so that the data volume can be reduced, the transmission efficiency can be improved, and the communication quality of the whole communication system can be ensured.

The communication system provided by the embodiment of the application can ensure the characteristics of reliable communication quality and flexible deployment of the original wireless intercom system, and simultaneously utilizes the characteristics and flexibility of wide-range coverage of internet communication, so that users far away from the wireless intercom system can also access the wireless intercom system to realize group communication, and the problem that the number of users in the communication system is limited due to the limitation of wireless channel resources of the wireless intercom system can be solved.

For example, in some application scenarios, if the number of users needing to access the call system is large, the wireless communication channel resources are limited, so that the terminal device 14 is also limited to access, and in order to access more users, some users may select to access the call system through the client 13, so as to greatly increase the number of users accessing the call system. In addition, the communication quality between the terminal device 14 and the host device 11 is high because the communication can be performed through independent frequency bands, and the communication quality between the client 13 and the host device 11 is slightly poor because the communication is performed through the internet. Therefore, when the communication system is used for realizing the cooperative work of the workers in the activity site, the terminal device 14 can be selected to be used for accessing the roles which are more important in the workers and are within the wireless communication coverage range, and the client 13 can be selected to be used for accessing the roles which are less important in the workers or are not within the wireless communication coverage range, so that the communication system can accommodate more users, the influence on the communication quality can be reduced as much as possible, and the smooth progress of the activity can be ensured.

In some embodiments, when encoding the mixed signal corresponding to each client, the host device 11 may perform compression encoding on the mixed signal by using a Low-latency and Low-loss encoding technique, for example, the Low-latency and Low-loss encoding technique may include one or more of an OPUS encoding technique, a Speex encoding technique, an ILBC (Internet Low bit code) encoding technique, an ISAC (Internet Speech Audio code) encoding technique, and a SILK encoding technique. Therefore, the call delay can be reduced as much as possible, and the call quality cannot be greatly influenced.

The number of terminal devices 14 in the call system is limited due to the limited radio channel resources for wireless communication between the terminal device 14 and the host device 11. While the clients 13 and the host device 11 are connected through internet resources, the number thereof is not limited too much, and thus, a plurality of clients 13 can be accessed into the call system, so that the number of users in the call system can be increased. Similarly, in order to ensure that the plurality of clients 13 can still work normally when being connected to the call system, the host device 11 may include a plurality of codec channels, each codec channel corresponds to one client 13, and is configured to decode the received second voice signal sent by the client 13, perform audio mixing processing, encode the audio mixing signal sent to the client 13, and send the encoded audio mixing signal to the cloud service end 12.

For example, as shown in fig. 3, the client 13 may include a client 1, a client 2, a client 3, and the like, and correspondingly, the host device 11 may include a coding/decoding channel 1, a coding/decoding channel 2, a coding/decoding channel 3, and the like corresponding to the client, where the coding/decoding channel 1 is configured to decode a second voice signal 1 acquired by a mobile terminal where the client 1 is located, so as to obtain a mixed sound signal 1 corresponding to the client 1 by using the second voice signal and the first voice signal through decoding, and then encode the mixed sound signal 1 by using the coding/decoding channel 1, and after obtaining the encoded mixed sound signal 1, forward the encoded mixed sound signal 1 to the client 1 through the cloud server. The voice signals of the other clients 2 and 3 can be encoded and decoded by using the encoding and decoding channels 2 and 3, respectively.

In some embodiments, since the host device 11 and the cloud server 12 are connected via the internet, and the cloud server 12 and the client 13 are also connected via the internet, before forwarding the mixed-sound signal corresponding to each client 13 to the client 13 via the cloud server 12, the host device 11 may encapsulate the mixed-sound signal based on a Voice Over Internet Protocol (VOIP), and then send the encapsulated mixed-sound signal to the cloud server 12, so that the cloud server forwards the mixed-sound signal to the client 13, and plays the packaged mixed-sound signal to the user via the client 13.

In some embodiments, as shown in fig. 4, the host device 11 may include a communication module 111, a coding/decoding module 112, and a mixing processing module 113, where the communication module 111 is configured to receive the first voice signal and the encoded second voice signal, transmit the received first voice signal to the mixing processing module 112, transmit the received encoded second voice signal to the coding module 112, receive and send a mixing signal corresponding to one of the terminal devices 14 to the terminal device 14, which is transmitted by the mixing processing module 112, and receive and send a mixing signal corresponding to one of the clients 13 to the client 13, which is transmitted by the mixing processing module 112.

For example, since the host device 11 is wirelessly connected to the terminal device 14, the communication module 111 at least includes a wireless communication sub-module for transmitting voice signals with the terminal device 14. Meanwhile, the host device 11 and the cloud service end 12 may be in communication connection through WiFi, a mobile communication network (e.g., 4G or 5G), a telecommunication network, and the like, and thus, the communication module 111 may also include a sub-module for implementing the communication connection. The communication module may be a chip for implementing a communication function, for example, a chip for implementing a wireless communication function, a chip for implementing a wired communication function, or a combination of the two. In some embodiments, the communication module may also be a CPU of the host device 11, i.e. the CPU may have communication functions integrated therein.

The encoding and decoding module 112 is configured to decode the received encoded second speech signal, transmit the decoded second speech signal to the audio mixing processing module 113, receive the audio mixing signal corresponding to one of the clients 13 and transmitted by the audio mixing processing module 113, perform encoding processing on the received audio mixing signal, send the encoded signal to the cloud server 12, and forward the encoded signal to the client 13 through the cloud server 12. Since there is a delay in the transmission of the voice signal through the internet, in order to reduce the delay as much as possible, improve the stability of the voice signal transmitted through the internet, and ensure the voice quality, the host device 11 may further include an encoding/decoding module 112 for performing compression encoding processing on the audio mixing signal sent to the client 13 and then sending the audio mixing signal. Similarly, the client 13 may also compress and encode the collected second speech signal and then transmit the encoded second speech signal to the host device 11, so that the host device 11 may decode the second speech signal and then perform mixing processing. The codec module 112 may be a chip or a processor with codec capability.

In some embodiments, the codec module 112 of the host device 11 may be a CPU of the host device 11, that is, the CPU is configured to implement various interface management, and network extension functions of the host device 11, and also perform codec functions.

Of course, if the CPU of the host device 11 is directly utilized to perform encoding and decoding, the computing power of the CPU is limited, and the data of the encoding and decoding channels on the CPU is also limited, and if the number of the clients 13 to be accessed is increased and the number of the required encoding and decoding channels is increased, there is a problem that more clients 13 cannot be added due to the limited computing power of the CPU. To solve the above problem, the codec module 112 may also be a daughter card dedicated to codec and connected to the CPU, and the daughter card is connected to the CPU of the host device 11 through a hardware interface. The sub-card may be a processor or a dedicated chip with codec capability, and multiple hardware interfaces (e.g., PCIE interfaces) may be reserved on the CPU for accessing the sub-card.

In some embodiments, the number of daughter cards is based on the number of clients 13 that may be desired to be accessed. For example, the number of codec channels in each daughter card is fixed, and may be one or more, and each client 13 needs one codec channel, so that the larger the number of clients 13 accessed, the larger the number of daughter cards.

The coding and decoding are carried out through the daughter card connected with the CPU, and the number of the coding and decoding channels can be adjusted by adjusting the number of the daughter cards connected with the CPU, so that the requirement of different numbers of clients can be met, and the method is more convenient and flexible.

The mixing processing module 113 is configured to perform mixing processing on the first voice signal and the second voice signal obtained by decoding, and generate a mixing signal corresponding to one of the terminal devices 14 and a mixing signal corresponding to one of the clients 13. The mixing processing module 113 may be a chip or a processor integrated with mixing processing functions.

In some embodiments, the mixing processing module 113 may be a DSP chip or an FPGA chip. For example, a DSP chip or an FPGA chip may be used to perform mixing processing on the voice signals of the members in a group. In addition, since the call system is further connected to the client 13, that is, the audio mixing processing module 113 needs to perform audio mixing processing on voices of more users, in order to meet the requirement that the call system can accommodate more users, a DSP chip or an FPGA chip with higher computing power can be properly selected.

In some embodiments, the voice signal transmitted between the mixing processing module 113 and the coding/decoding module 112 may be a Pulse Code Modulation (PCM) voice signal, for example, a plurality of PCM channels may be included between the mixing processing module 113 and the coding/decoding module 112 for transmitting a lossless second voice signal and a mixing signal.

In some implementations, the plurality of end devices 14 and the plurality of clients 13 may be divided into a plurality of groups, and full-duplex calls may be performed between the clients 13 and the end devices 14 in the same group, and calls may be isolated between different groups. In order to support multiple talk groups, the mixing processing module 113 may also include multiple mixing processing modules, where each mixing processing module 113 corresponds to a group and is configured to perform mixing processing on a first voice signal sent by a terminal device 14 and a second voice signal sent by a client 13 in the group.

For example, the staff members in the activity process are usually divided into a plurality of groups according to their duties, and the staff members in each group can implement voice call. Taking a movie recording activity as an example, the staff members during the activity can be divided into a camera group, a light group, a clothes group, etc., so that the terminal devices 14 or the clients 13 used by the staff members can be divided into the above three groups. The mixing processing module 113 may perform mixing processing on the voice signals of the members of each group respectively to send the mixed signals to the members in the group. For example, before performing the group call, the user may perform group management and configuration on the terminal device 14 and the client 13 (for example, through an interactive interface on the host device 11), and divide the terminal device 14 and the client 13 into a plurality of groups, where both the terminal device 14 and the client 13 may uniquely identify their own identification information, and the user may group them based on the identification information.

In some embodiments, the number of end devices 14 and the number of clients 13 are divided into a plurality of groups, and the members in each group may implement a full-duplex call. Each client 13 needs to join the talk group formed by the terminal devices, and then can realize the talk with other members in the group. Therefore, before sending the second voice signal to the host device 11 through the cloud server 12, each client 13 may also send a request for joining a target group to the cloud server 12, where the request carries a group identifier of the target group, and the group identifier is used to uniquely identify the target group.

In some embodiments, the cloud server 12 may have a group management function, and thus, the cloud server 12 may directly add each client 13 to the target group based on the group identification of the target group. In some embodiments, the cloud server 12 may not have the function of group management, and thus, the cloud server 12 may forward the request to the host device 11, so that the host device 11 may add each client 13 to the target group based on the group identification of the target group.

For example, when a group is managed and assigned by a user, each time a group is created, the group has a group identifier that uniquely identifies the group, which may be, for example, a string of numbers or letters, or a combination of the two. When a user wants to access a group of the call system through a mobile terminal equipped with a client 13, the user may first determine a group identifier of the group, and then initiate a request carrying the group identifier through the client, after receiving the request, the cloud server 12 may add the client to the corresponding group based on the group identifier, or the cloud server 12 forwards the request to the host device 11, and the host device 11 adds the client to the corresponding group, so that the user may perform a call based on the client 13 and the terminal device 14 or the client 13 in the group.

In some embodiments, the communication system further includes a plurality of terminal devices 14, where the plurality of terminal devices 14 may be connected to a device with voice collecting and playing functions (such as a headset), and configured to send a first voice signal collected by the device with voice collecting and playing functions to the host device 11, and receive a mixed-sound signal sent by the host device 11 and play the mixed-sound signal through the device with voice collecting and playing functions.

In some embodiments, in order to increase the coverage area and the communication distance of the terminal devices 14 and increase the number of the terminal devices 14, as shown in fig. 5, the communication system 10 further includes a plurality of wireless forwarding devices 16, each wireless forwarding device 16 is wirelessly connected to a plurality of terminal devices 14, and each wireless forwarding device 16 is connected to the host device 11 through the ethernet network, so as to form a wireless access network with a larger coverage area. The wireless forwarding device 16 is configured to receive the first voice signals sent by the terminal devices 14 and forward the first voice signals to the host device 11, and receive mixed sound signals sent by the host device 11 and corresponding to each terminal device 14 and forward the mixed sound signals to each terminal device 14. The host device 11 connects the wireless forwarding devices 16 in the wireless access networks in different frequency bands, so that the voice signal received by the wireless forwarding device 16 from the terminal device 14 can be forwarded to another wireless forwarding device 16 through the host device 11, and the terminal device 14 connected with any one wireless forwarding device 16 can communicate with the terminal devices 14 connected with other wireless forwarding devices 16, so that the number of the terminal devices 14 capable of communicating at the same time is increased, the capacity expansion of the terminal devices 14 is realized, and the communication distance and area are also enlarged.

In some embodiments, each client 13 is an APP installed on the mobile terminal. For example, the mobile phone can be a voice call software installed on the mobile phone. The user may download the installation program of the session software from the cloud service end 12, install the installation program, and then join the session group formed by the terminal devices 14 by using the session software. The APP supports access to a call system, realizes communication with the terminal device 14, and meanwhile, the APP may also have functions of encoding and decoding, echo cancellation, receiving and transmitting of VOIP signals, and the like.

In the related art, when the call system is managed, the call system is usually managed through an interactive interface or a key on the host device 11, and the management method is single, and the user needs to walk to the location of the host device 11 to operate, which is cumbersome and inflexible. In some embodiments, in order to implement more convenient and flexible management of the call system, the management mode of the call system may be extended, that is, the cloud server 12 may provide management functions such as network configuration, user management, group setting, and voice parameter adjustment for the call system, and the dedicated client 13 with management authority may perform remote management on the whole call system at any time, so that the management mode is more flexible and changeable. Therefore, in some embodiments, the designated client 131 in the plurality of clients 13 is further configured to send a management request to the cloud server 12, the cloud server 12 is further configured to perform a corresponding management operation on the call system based on the management request, or the cloud server may send the management request to the host device 11, so that the host device 11 performs a corresponding management operation on the call system.

In some embodiments, the management operation includes one or more of: group management operation, operation of configuring voice parameters of a call system, and operation of adding a registered user are performed for a plurality of clients 13 or a plurality of terminal devices 14. That is, the user can manage the group members by operating on the interface of the designated client 131 with the management authority, for example, dividing the client 13 or the terminal device 14 in the call system into different groups, or adding a new member to an existing group, or deleting an existing member in the group. Or may configure voice parameters of the telephony system by specifying the client 131, or add a newly registered user to the telephony system, etc. By adding a mode of managing the call system by the cloud service end, the management of the call system is more flexible and changeable.

In some embodiments, the cloud service end 12 is further configured to mix second voice signals received from the plurality of clients 13 into one path of voice signal and send the one path of voice signal to the host device 11, and the transmission efficiency of the second voice signal can be improved by combining multiple paths of second voice signals collected by the clients 13 into one path in the cloud service end 12 and sending the one path of second voice signal. Certainly, a certain time delay is also brought by adding the step of mixing sound at the cloud service end 12, so that whether to deploy mixing sound at the cloud service end can be comprehensively determined based on the improvement range of the transmission efficiency and the time delay brought by mixing sound.

To further explain the call system provided in the embodiments of the present application, the following explanation is made with reference to a specific embodiment.

As shown in fig. 6, which is a schematic diagram of a call system in an embodiment of the present application, the call system includes a plurality of waist packs 21, each waist pack 21 is connected to a headset 22, one or more remote call units 23 (such as remote call unit a and remote call unit B in the figure), and each remote call unit 23 is located in a local area network and is wirelessly connected to one or more waist packs 21. The conversation remote unit 23 is connected with the host device 24 through the ethernet, the host device 24 is connected with the cloud server 25 provided with the conversation service based on the internet protocol, and the cloud server 25 is connected with the mobile terminal 26 provided with the conversation APP based on the internet protocol.

The user can divide the call belt pack 21 and the mobile terminal 26 into a plurality of groups, and members in each group can perform a group call. For example, the belt pack 21 and the mobile terminal 26 in the figure can be divided into 3 different groups, group 1, group 2 and group 3, where:

group 1, A1, A2, B1, B5, M1;

group 2, A3, A4, A6, A7, B2, B3, M2;

group 3, A5, A8, B4, B6, B7, B8, M3.

The headset 22 can output the voice signal of the user to the call belt pack 21 after acquiring the voice signal, and the call belt pack is a wireless terminal device powered by a battery, and can realize wireless voice receiving and sending in the call process. The talk belt pack 21 can transmit the voice signal collected by the headset 22 to the talk remote unit 23, the talk remote unit 23 has a wireless transceiving function, and an ethernet extension function, and can convert the voice signal transmitted by the accessed wireless belt pack 21 into a digital IP signal and transmit the voice signal to the host device 24 through a voice transmission protocol based on the internet. The transmitted voice signal between the headset 22 and the host device 24 may be a PCM (pulse code modulation) signal, among others.

The mobile terminal 26 may collect a user voice signal, and then transmit the user voice signal to the cloud server 25 through the internet after performing OPUS coding on the voice signal through the call APP installed on the mobile terminal 26, so that the call service provided on the cloud server 25 forwards the OPUS coded voice signal to the host device 24.

The internal structure of the host device 24 is as shown in fig. 7, and at least includes a CPU and a DSP/FPGA chip, the CPU can run in a Linux OS system, and in some scenarios, the CPU can integrate an OPUS encoding and decoding function, and is configured to decode and process a received voice signal collected by the mobile terminal 26 and forwarded by the cloud server 25, and then send the decoded voice signal to the DSP/FPGA chip for audio mixing processing. And is configured to perform OPUS encoding on the mixed sound signal after the mixed sound processing, and send the mixed sound signal to the cloud server 25. Since the mobile terminal 26 includes a plurality of OPUS codec channels, the CPU may include a plurality of OPUS codec channels, each OPUS codec channel is used for performing codec processing on a voice signal of one mobile terminal 26. Of course, since the number of OPUS codec channels is limited by the CPU computing power, when the number of mobile terminals 26 accessed is large, a single CPU computing resource cannot support the access of these multiple mobile terminals 26. In some scenarios, the CPU may also include multiple hardware interfaces on which the user accesses a daughter card dedicated to encoding and decoding, through which OPUS encoding and decoding is implemented. The number of the OPUS coding and decoding channels contained in each daughter card is fixed, and the coding and decoding functions of the CPU can be expanded by inserting a plurality of daughter cards into the CPU, so that the requirement that the number of the accessed mobile terminals 26 is different is met. Moreover, the CPU Linux OS also supports a TCP/IP protocol stack and a Voice Over Internet Protocol (VOIP) transmission protocol, and may encapsulate the audio mixing signal based on the VOIP transmission protocol, transmit the audio mixing signal to the cloud server 25, and forward the audio mixing signal to the mobile terminal 26 through the cloud server 25.

The DSP/FPGA chip may integrate a sound mixing processing function for performing sound mixing processing on a voice signal received by the host device 24, for example, a voice signal sent by the wireless belt pack 21, a voice signal collected by a local headset, a two-wire analog voice signal or a voice signal collected by the mobile terminal 26 forwarded by the cloud server 25. Through sound mixing processing, a sound mixing signal corresponding to each call user in the group is generated, for example, a sound mixing signal corresponding to each call belt pack 21 is generated, the sound mixing signal is forwarded to the call belt pack 21 through the wireless remote unit 23, a sound mixing signal corresponding to each mobile terminal 26 is generated, then, after the OPUS coding is performed through an integrated OPUS coding and decoding module on the CPU, an OPUS coding code stream is obtained, and then, the obtained OPUS coding code stream is forwarded to the mobile terminal 26 through the cloud server 25. The group can comprise a plurality of groups, and the DSP/FPGA chip can be used for respectively processing the sound mixing of the voice signals of the members in each group. The number of groups can be configured according to the processing capacity of the DSP/FPGA chip and the field application requirement. The voice signal transmitted between the CPU and the DSP/FPGA can be a PCM voice signal.

The cloud server 25 may be a server or a cluster of a mainstream cloud service provider, such as amazon cloud, microsoft cloud, airy cloud, or Tencent cloud, and the main user forwards a voice signal sent by the mobile terminal 26 to the host device 24 and forwards a mixed sound signal after being mixed by the host device 24 to the mobile terminal 26, and meanwhile, the cloud server may also deploy service functions of user registration management, group allocation, voice parameter configuration, wireless in-line system management, and the like for the whole call system, and these functions may be flexibly invoked and controlled through an APP on the mobile terminal 26 having a management authority.

The APP on the mobile terminal 26 can support call voice access, voice processing functions, such as AEC echo cancellation, noise reduction, tone quality improvement and the like, OPUS voice coding and decoding, network VOIP stream receiving and sending and the like, the mobile intelligent terminal with the cloud call APP has the conventional function of a wireless call waist pack, and the accessible number and the access authority are managed by a cloud call service network side.

In addition, the host device 24 may also include an ethernet interface for connecting with the wireless remote unit 23 via ethernet, or for connecting with other host devices 24 via ethernet. A Tally device interface may also be included in the host device 24 for interfacing with a Tally device. In addition, the host device 24 may also set keys, LCD screens, etc. based on actual needs.

The call system of the embodiment has the following advantages:

(1) The communication system provided by the embodiment can ensure the characteristics of reliable communication quality and flexible deployment of the original wireless intercom system, and simultaneously utilizes the characteristics and flexibility of wide-range coverage of internet communication, so that users far away from the wireless intercom system can also access the wireless intercom system to realize group communication.

(2) The mobile terminal APP can be accessed into the call system through the Internet, so that the problem that the number of users in the call system is limited due to the limitation of wireless channel resources of a wireless intercom system can be solved, namely, the user capacity of the call system is increased. The newly-added mobile terminal APP can realize wide area access equivalent to a special wireless waist pack, and flexibility and expandability are greatly enhanced.

(3) The management of the whole call system can be realized through the mobile terminal APP with the management authority, for example, group management, voice parameter configuration, user registration and the like, and the management mode of the call system is increased, so that the management of the call system is more flexible and changeable.

Further, the embodiment of the application also provides a wireless host device, wherein the host device is in communication connection with the plurality of terminal devices; the host equipment and the plurality of clients respectively establish a connecting channel with the cloud server through the Internet; the host device comprises a communication module, a coding and decoding module and a sound mixing processing module,

the communication module is configured to receive a first voice signal sent by the terminal device, receive a second encoded voice signal sent by the client and forwarded by the cloud server, transmit the first voice signal to the audio mixing processing module, transmit the second encoded voice signal to the encoding and decoding module, receive an audio mixing signal corresponding to one of the terminal devices and transmitted by the audio mixing processing module, and receive an audio mixing signal corresponding to one of the client and transmitted by the audio mixing processing module and transmitted to the client;

the encoding and decoding module is used for decoding the received encoded second voice signal and then transmitting the decoded second voice signal to the audio mixing processing module, receiving the audio mixing signal which is transmitted by the audio mixing processing module and corresponds to one of the client sides, and encoding the audio mixing signal so as to send the encoded audio mixing signal to the cloud server side;

The specific structure and function of the host device may refer to the description in the above embodiments, and are not described herein again.

In addition, the embodiment of the application also provides a communication system, which comprises a host device, a plurality of terminal devices, a cloud server and a plurality of clients, wherein the host device is in communication connection with the terminal devices; the host equipment and the client establish a connection channel with the cloud server through the Internet respectively;

the client is used for acquiring a second voice signal, encoding the second voice signal and then sending the second voice signal to the cloud server;

The specific structure and function of the communication system may refer to the description in the foregoing embodiments, and are not described herein again.

The various technical features in the above embodiments can be arbitrarily combined, so long as there is no conflict or contradiction between the combinations of the features, but the combination is limited by the space and is not described one by one, and therefore, any combination of the various technical features in the above embodiments also falls within the scope disclosed in the present specification.

Other embodiments of the present disclosure will be apparent to those skilled in the art from consideration of the specification and practice of the specification disclosed herein. The embodiments of the present specification are intended to cover any variations, uses, or adaptations of the embodiments of the specification following, in general, the principles of the embodiments of the specification and including such departures from the present disclosure as come within known or customary practice in the art to which the embodiments of the specification pertain. It is intended that the specification and examples be considered as exemplary only, with a true scope and spirit of the embodiments being indicated by the following claims.

It is to be understood that the embodiments of the present specification are not limited to the precise arrangements described above and shown in the drawings, and that various modifications and changes may be made without departing from the scope thereof. The scope of the embodiments of the present specification is limited only by the appended claims.

The above description is only for the purpose of illustrating the preferred embodiments of the present disclosure, and is not intended to limit the embodiments of the present disclosure, and any modifications, equivalents, improvements, etc. made within the spirit and principle of the embodiments of the present disclosure should be included in the scope of the embodiments of the present disclosure.

Claims

1. The conversation system is characterized by comprising a host device, a cloud server and a plurality of clients, wherein the host device is in communication connection with a plurality of terminal devices; the host equipment and the client establish a connection channel with the cloud server through the Internet respectively;

2. The communication system according to claim 1, wherein the host device is further configured to perform encoding processing on the audio-mixed signal corresponding to the one of the clients based on a low-loss and low-delay encoding technique, and forward the encoded audio-mixed signal to the client through the cloud server; the low-loss and low-delay coding technology comprises one or more of an OPUS coding technology, a Speex coding technology, an ILBC coding technology, an ISAC coding technology and a SILK coding technology.

3. The communication system according to claim 1 or 2, wherein the host device includes a plurality of codec channels, each codec channel corresponds to one of the clients, and is configured to decode the encoded second voice signal sent by the client, perform audio mixing processing on the decoded second voice signal and the first voice signal, encode the audio mixing signal corresponding to the client, and forward the encoded audio mixing signal to the client through the cloud server.

4. The communication system according to claim 1, wherein the host device is further configured to encapsulate the audio mixing signal corresponding to the one of the clients based on an internet voice transfer protocol and send the encapsulated audio mixing signal to the cloud server.

5. The communication system of claim 1, wherein the host device comprises a communication module, a codec module, and a mixing module,

the communication module is configured to receive the first voice signal and the encoded second voice signal, transmit the first voice signal to the audio mixing processing module, transmit the encoded second voice signal to the encoding and decoding module, receive the audio mixing signal corresponding to the one terminal device and transmitted by the audio mixing processing module, and receive the audio mixing signal corresponding to the one client and transmitted by the audio mixing processing module and transmitted to the client;

and the sound mixing processing module is used for carrying out sound mixing processing on the first voice signal and the second voice signal obtained by decoding to generate a sound mixing signal corresponding to one of the terminal devices and a sound mixing signal corresponding to one of the clients.

6. The telephony system of claim 5, wherein the codec module is a CPU of the host device, or wherein the codec module is a daughter card connected to the CPU of the host device via a hardware interface.

7. The telephony system of claim 6, wherein the number of daughter cards is determined based on the number of clients.

8. The communication system according to claim 5, wherein the mixing processing module is a DSP chip or an FPGA chip.

9. The communication system according to claim 5, wherein the plurality of terminal devices and the plurality of clients are divided into a plurality of groups, and the audio mixing processing module is configured to perform audio mixing processing on a first audio signal sent by the terminal device and a second audio signal sent by the client in each group.

10. The communication system according to claim 5, wherein the voice signal transmitted between the mixing processing module and the encoding/decoding module is a pulse code modulation voice signal.

11. The communication system according to claim 1, further comprising a plurality of terminal devices, wherein each terminal device is connected to a voice collecting and playing device, and configured to send the first voice signal collected by the voice collecting and playing device to the host device, and receive the audio mixing signal sent by the host device and play the audio mixing signal through the voice collecting and playing device.

12. The telephony system of claim 1, further comprising a plurality of wireless forwarding devices, each wireless forwarding device being wirelessly coupled to a plurality of end devices, each wireless forwarding device being coupled to the host device via Ethernet,

the wireless forwarding equipment is used for receiving the voice signals collected by the plurality of terminal equipment, forwarding the voice signals to the host equipment, receiving the sound mixing signals sent by the host equipment and sent by each terminal equipment, and forwarding the sound mixing signals to each terminal equipment.

13. The telephony system of claim 1, wherein each client is an APP installed on a mobile terminal.

14. The communication system of claim 1, wherein the number of terminal devices and the number of clients are divided into a plurality of groups,

before sending the encoded second voice signal to the host device through the cloud server, each client is further configured to send a request for joining a target group to the cloud server, where the request carries a group identifier of the target group;

the cloud server is configured to add the each client to the target group based on the group identifier of the target group, or the cloud server forwards the request to the host device, so that the host device adds the each client to the target group based on the group identifier of the target group.

15. The telephony system of claim 1, wherein a designated client of the plurality of clients is further configured to send a management request to the cloud server, and the cloud server is further configured to perform a corresponding management operation on the telephony system based on the management request, or the cloud server is further configured to forward the management request to the host device, so that the host device performs a corresponding management operation on the telephony system.

16. The telephony system of claim 15, wherein the management operations include one or more of: the operation of group management, the operation of configuring the voice parameters of the call system and the operation of adding registered users are carried out on the plurality of clients or the plurality of terminal devices.

17. The communication system according to claim 1, wherein the cloud server is further configured to receive encoded second voice signals sent by the plurality of clients, mix the encoded second voice signals into one path of voice signal, and send the path of voice signal to the host device.

18. A host device, wherein the host device is communicatively coupled to a plurality of terminal devices; the host equipment and the plurality of clients respectively establish a connection channel with the cloud server through the Internet; the host device comprises a communication module, a coding and decoding module and a sound mixing processing module,

19. The host device according to claim 18, wherein the codec module is further configured to encode the audio-mixed signal corresponding to the one of the clients based on a low-loss and low-delay encoding technique, and forward the encoded audio-mixed signal to the client through the cloud server; the low-loss low-latency encoding technique includes an OPUS encoding technique.

20. The host device according to claim 18 or 19, wherein the codec module includes a plurality of codec channels, each codec channel corresponds to one of the clients, and is configured to decode a second voice signal sent by the client and then perform mixing processing on the second voice signal and the first voice signal, and encode a mixing signal corresponding to the client and then forward the mixing signal to the client through the cloud server.

21. The host device of claim 18, wherein the communication module is further configured to encapsulate the mixed-sound signal corresponding to the one of the clients according to a voice over internet protocol and send the encapsulated mixed-sound signal to the cloud server.

22. The host device of claim 18, wherein the codec module is a CPU of the host device or a daughter card connected to the CPU of the host device via a hardware interface.

23. The host device of claim 22, wherein the number of daughter cards is determined based on the number of clients.

24. The host device of claim 18, wherein the mixing processing module is a DSP chip or an FPGA chip.

25. The host device according to claim 18, wherein the plurality of terminal devices and the plurality of clients are divided into a plurality of groups, and the mixing processing module is configured to perform mixing processing on a first voice signal sent by the terminal device and a second voice signal sent by the client in each group.

26. A call system is characterized by comprising a host device, a plurality of terminal devices, a cloud server and a plurality of clients, wherein the host device and the terminal devices are in the same wireless access network; the host equipment and the client establish a connection channel with the cloud server through the Internet respectively;