Call system, call host, and call terminal
By dividing the call terminal into call packets and performing mixing and silencing in the wireless internal call system, the problem of limited number of call terminals and large battery consumption in the call system is solved, and higher time slot utilization and longer battery life time are achieved.
Patent Information
- Application Number
- PCT/CN2023/114218
- Authority / Receiving Office
- WO · WO
- Patent Type
- Applications
- Current Assignee / Owner
- Filing Date
- 2023-08-22
- Publication Date
- 2025-02-27
AI Technical Summary
In the existing wireless internal calling system, the number of call terminals is limited, and the number of call users cannot be increased. The battery capacity of the call host consumes faster and the working time is limited.
By dividing multiple call terminals into one or more call packets, the call host mixes the voice signals sent by the call terminals in each packet, obtains a unified mixing signal, and broadcasts to each call terminal in the packet through a downlink time slot. After receiving the mixing signal, the call terminal performs silence processing to eliminate the voice signal collected by itself.
This improves the utilization rate of time slots, frees up more time slots to be used as uplink time slots, increases the number of call users in the call system, saves power consumption of the call host, and extends the battery life time.
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Figure CN2023114218_27022025_PF_FP_ABST
Abstract
Description
Call system, call host and call terminal Technical Field
[0001] This specification relates to the field of communication technology, and in particular to a call system, a call host, and a call terminal. Background Art
[0002] In scenes like film shoots, stage performances, and large-scale events, wireless intercom systems are often needed to facilitate communication between on-site staff and facilitate work organization and scheduling. Typically, a wireless intercom system consists of a host computer and multiple terminals (such as beltpacks) that can wirelessly connect to the host computer. The terminals can transmit collected user voice signals to the host computer, which then mixes the received voice signals. Specifically, for each terminal, it mixes the voice signals sent by all other terminals except that terminal and then forwards them to that terminal.
[0003] To ensure that each call terminal can operate in full-duplex call mode, these multiple call terminals usually operate in different communication time slots using a time-division method. Since each call terminal sends different voice signals to the call host, each call terminal needs to occupy an uplink time slot. The mixed audio signal sent by the call host to each call terminal is also different, so each call terminal also needs to occupy a downlink time slot. Since the number of time slots in each data transmission frame is limited, each call terminal needs to occupy two time slots, resulting in a limited number of call terminals supported by a call system, and the number of call users cannot be increased. In addition, since each call terminal needs to occupy a downlink time slot, as the number of call terminals increases, the call host's transmission power consumption will increase. Therefore, when the number of call terminals is large, the call host's battery power will be consumed faster, and the operating time will be limited.
[0004] Summary of the Invention
[0005] Based on this, the embodiments of this specification provide a call system, a call host, and a call terminal.
[0006] According to a first aspect of an embodiment of this specification, a call system is provided, comprising a call host and a plurality of call terminals, wherein the call host is wirelessly connected to the plurality of call terminals, respectively, and the plurality of call terminals are divided into one or more call groups;
[0007] At least some of the multiple call terminals are used to collect user voice signals and send the voice signals to the call host;
[0008] The call host is used to mix the voice signals sent by the call terminals in each call group to obtain a mixed signal, and send the mixed signal to each call terminal in the call group;
[0009] At least some of the call terminals are further configured to perform silencing processing on the received mixed audio signal to eliminate the voice signal collected by the terminals themselves, and to play the mixed audio signal after the silencing processing.
[0010] According to a second aspect of an embodiment of this specification, a call host is provided, wherein the call host is communicatively connected to a plurality of call terminals, and the plurality of call terminals are divided into one or more call groups;
[0011] The call host is used to receive voice signals collected by at least some of the multiple call terminals, and for each call group, mix the voice signals sent by the call terminals in the call group to obtain a mixed signal, and send the mixed signal to each call terminal in the call group, so that at least some of the call terminals are also used to mute the received mixed signal to eliminate the voice signals collected by themselves, and play the mixed signal after mute processing.
[0012] According to a third aspect of the embodiments of this specification, there is provided a call terminal, the call terminal being communicatively connected to a call host;
[0013] The call terminal is used to collect the user's voice signal and send the voice signal to the call host so that the call host mixes the received voice signal to obtain a mixed signal; and, receives the mixed signal sent by the call host, performs silencing on the received mixed signal to eliminate the voice signal collected by itself, and plays the mixed signal after silencing.
[0014] Using the solutions of the embodiments of this specification, for a single call group, after receiving voice signals from each call terminal within that group, the call host can mix the voice signals sent by all call terminals within the group to generate a unified mixed signal. This mixed signal can then be broadcast to each call terminal within the group via a single downlink timeslot. Since the mixed signal received by a call terminal may include its own collected voice signal, the call terminal can mute the mixed signal after receiving it, eliminating its own collected voice signal. This allows the call terminal to only play the voice signals collected by other call terminals within the group. Compared to allocating a separate downlink timeslot to each call terminal, this approach allows call terminals within the same group to share a single downlink timeslot, improving timeslot utilization and freeing up more timeslots for uplink use. This allows each call host to connect to more call terminals, increasing the number of call users in the call system. Furthermore, since the call host only needs to transmit one mixed signal for each call group, it also saves power and extends battery life.
[0015] It should be understood that the foregoing general description and the following detailed description are exemplary and explanatory only and are not restrictive of the present disclosure. BRIEF DESCRIPTION OF THE DRAWINGS
[0016] The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the specification and, together with the description, serve to explain the principles of the specification.
[0017] FIG1 is a schematic diagram of the mixing logic of a call host of a wireless call system in the prior art.
[0018] FIG2 is a schematic diagram of time slot allocation in a wireless communication system in the prior art.
[0019] FIG3 is a schematic diagram of the structure of a communication system according to an embodiment of the present specification.
[0020] FIG4 is a schematic diagram of a call terminal performing a noise cancellation process according to an embodiment of the present specification.
[0021] FIG5 is a schematic diagram of the structure of a communication system according to an embodiment of the present specification.
[0022] FIG6 is a schematic structural diagram of a communication system according to an embodiment of the present specification.
[0023] FIG7 is a schematic structural diagram of a communication system according to an embodiment of the present specification. DETAILED DESCRIPTION
[0024] Exemplary embodiments will be described in detail herein, with examples illustrated in the accompanying drawings. In the following description, when referring to the drawings, identical numerals in different figures represent identical or similar elements unless otherwise indicated. The embodiments described in the following exemplary embodiments are not intended to represent all embodiments consistent with this specification. Rather, they are merely examples of apparatus and methods consistent with certain aspects of this specification, as detailed in the appended claims.
[0025] The terms used in this specification are for the purpose of describing specific embodiments only and are not intended to limit this specification. As used in this specification and the appended claims, the singular forms "a," "an," "the," 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.
[0026] It should be understood that although the terms first, second, third, etc. may be used in this specification to describe various information, such information should not be limited to these terms. These terms are merely used to distinguish information of the same type from one another. For example, first information may also be referred to as second information, and similarly, second information may also be referred to as first information without departing from the scope of this specification. Depending on the context, the term "if" as used herein may be interpreted as "when," "when," or "in response to determining."
[0027] In scenes like film shoots, stage performances, and large-scale events, wireless intercom systems are often needed to facilitate communication between on-site staff and facilitate work organization and scheduling. Typically, a wireless intercom system consists of a host computer and multiple terminals (such as beltpacks) that can wirelessly connect to the host computer. The terminals can transmit collected user voice signals to the host computer, which then mixes the received voice signals. Specifically, for each terminal, it mixes the voice signals sent by all other terminals except that terminal and then forwards them to that terminal.
[0028] Figure 1 illustrates the mixing logic of an existing wireless intercom system. Typically, multiple terminals wirelessly connected to a call base can be divided into one or more call groups, and terminals within each call group can communicate with each other. Assume a call group includes a call base (BS) and eight call terminals (PU1-PU8). Since each call terminal only needs to hear the voice signals collected by other call terminals within the same call group, it does not need to hear its own voice signals. Therefore, after each call terminal sends its own voice signal to the call base, the call base can mix the voice signals collected by other call terminals other than that terminal and then send them to that terminal. For example, the call base will receive voice signals from PU1-PU8 and mix the voice signals from PU2-PU7 with PU1. The mixed signal is then returned to PU1, which plays the mixed signal through its speaker, allowing the user using PU1 to hear the voice signals of other users in the group. Similarly, other call terminals can obtain their mixed signals using a similar method.
[0029] Typically, to ensure full-duplex communication for each endpoint, these multiple endpoints operate in different communication time slots using a time-division system. Because each endpoint transmits a different voice signal to the host, each endpoint occupies an uplink time slot. Furthermore, because the host sends a different mixed audio signal to each endpoint, each endpoint also occupies a downlink time slot. Since each data transmission frame has a limited number of time slots, each endpoint must occupy two time slots, limiting the number of endpoints supported within a single communication system.
[0030] Taking the example of communication between a call host and a call terminal using DECT (Digital Enhanced Cordless Telecommunications) technology, a complete DECT frame is 10ms long, which is evenly divided into 24 time slots. Figure 2 shows a schematic diagram of DECT time slot allocation, where the first 12 time slots are for the call host to send data and the call terminal to receive data, hereinafter referred to as downlink time slots. The last 12 time slots are for the call terminal to send data and the call host to receive data, hereinafter referred to as uplink time slots. As can be seen from Figure 2, each call terminal must occupy an uplink time slot to send its collected voice signal to the call host. At the same time, each call terminal must occupy a downlink time slot to receive the mixed signal from the call host, which is the result of mixing the voice signals of other call terminals.
[0031] Because each call terminal requires a separate uplink and downlink timeslot, time slot utilization is low. A standard DECT frame is divided into 24 timeslots. Excluding timeslots used for transmitting status information and handover (channel or frequency band switching), existing technology allows a single call center to support a maximum of 10 full-duplex wireless call terminals. Currently, large-scale events are increasing the number of call users supported by call systems. However, existing call systems only support a maximum of 10 full-duplex wireless call terminals, making this number of users unscalable and unable to support additional call demands. Furthermore, since each call terminal occupies a downlink timeslot, the increase in the number of call terminals also increases the call center's transmit power consumption. This means that the call center's power consumption increases with the number of call terminals. This shortens the battery life of the call center when there are a large number of call terminals.
[0032] Based on this, an embodiment of the present application provides a call system. Considering that in existing call systems, the call host needs to perform separate mixing processing for each call terminal to obtain the mixed signal corresponding to the call terminal, and therefore needs to allocate a separate downlink time slot for it to transmit the mixed signal, resulting in a low overall time slot utilization. Therefore, in an embodiment of the present application, the mixing logic of the call host is modified. For the same call group, after receiving the voice signals sent by each call terminal in the group, the call host can mix the voice signals sent by all call terminals in the group to obtain a unified mixed signal, which can then be broadcast to each call terminal in the group through a downlink time slot. Since the mixed signal received by the call terminal may include the voice signal collected by itself, and the call terminal does not need to play the voice signal collected by itself, after receiving the mixed signal, the call terminal can mute the mixed signal to eliminate the voice signal collected by itself, so that the call terminal can only play the voice signals collected by other call terminals in the group.
[0033] Compared to allocating a downlink timeslot to each call terminal individually, this approach allows terminals in the same group to share a single downlink timeslot, improving timeslot utilization and freeing up more uplink timeslots. This means each call center can connect to more terminals, increasing the number of users in the call system. Furthermore, since the call center only needs to transmit one mixed audio signal per call group, it also saves power and extends battery life.
[0034] As shown in FIG3 , a call system provided in an embodiment of the present application includes a call host 11 and multiple call terminals 12. The call host 11 is wirelessly connected to each of the multiple call terminals 12. For example, the call host 11 and the call terminals 12 can communicate wirelessly using short-range communication methods such as WiFi, Bluetooth, ZigBee, and DECT. Alternatively, to ensure call quality between the two, the call host 11 and the call terminals 12 can also communicate wirelessly using some customized proprietary protocols. For example, the call host 11 and the call terminals 12 can operate in independently allocated frequency bands, such as the UHF band, the 1.9 GHz DECT band, the 2.4 GHz / 5 GHz ISM band, etc.
[0035] Multiple call terminals 12 wirelessly connected to the call host 11 can be divided into one or more call groups based on actual needs. Calls can only be made between call terminals 12 within the same call group. When performing audio mixing, the call host 11 mixes only the voice signals collected by the call terminals 12 within the same call group and sends the mixed audio to all call terminals within that call group.
[0036] In some embodiments, each call terminal 12 may include an audio acquisition unit, an audio playback unit, an audio processing unit, and a wireless transmission unit. The audio acquisition unit can be used to collect the user's voice signal, for example, it can be a microphone. The audio playback unit can be used to play the voice signal, for example, it can be a speaker. The audio processing unit can be used to mute the mixed signal received from the call host 11 to eliminate the voice signal collected by itself. Among them, the audio processing unit can be a DSP chip, or an FPGA chip, etc. The wireless transmission unit can be used for wireless transmission and reception of data, and the wireless transmission unit can be various wireless transmission modules, etc. Of course, in some scenarios, the audio processing unit and the wireless transmission unit can also be integrated on the same chip.
[0037] Of course, the call terminal 12 can also be configured with other functional components according to actual functional requirements, and this embodiment of the application does not limit this.
[0038] Among them, at least some of the multiple call terminals 12 can be used to collect the user's voice signal and send the collected voice signal to the call host 11, as well as to mute the mixed signal received from the call host 11 to eliminate the voice signal collected by itself and play the mixed signal after mute processing.
[0039] For example, in some scenarios, the number of call terminals 12 in the call system is relatively small, so all call terminals 12 can be allocated uplink time slots. That is, all call terminals 12 can send collected voice signals to the call host, enabling both speaking and listening. In some scenarios, considering the limited number of communication time slots, it is impossible to allocate uplink time slots to each call terminal. However, some users just need to listen to the voices of other users in the group without speaking. Therefore, the call terminals 12 of these users do not need to be allocated uplink time slots, that is, they can only listen without speaking. In other words, only some of the multiple call terminals 12 connected to the call host 11 can achieve both speaking and listening.
[0040] The call host 11 can receive voice signals from the call terminal 12, and then for each call group, mix the voice signals sent by the call terminal 12 in the call group to obtain a mixed signal, and broadcast the mixed signal to each call terminal 12 in the call group.
[0041] The call host 11 may include a wireless transmission unit for wireless data transmission and reception. The call host 11 may also include a mixing processing unit for mixing voice signals. In some scenarios, the call host 11 may include multiple mixing processing units, each of which may be used to mix voice signals from call terminals 12 in a call group. The mixing processing units may be integrated into a DSP chip or an FPGA chip, with one or more mixing processing units being provided on the DSP chip or FPGA chip.
[0042] Of course, the call host 11 can also be configured with other functional components according to actual functional requirements, and this embodiment of the application does not limit this.
[0043] After collecting the voice signal, the call terminal 12 sends it to the call host 11 for mixing. The call terminal then receives the mixed signal processed by the call host 11. Before playing the mixed signal, the mixed signal can be muted to eliminate the voice signal collected by the call terminal 12. In some embodiments, to eliminate the voice signal collected by the call terminal 12 from the mixed signal, the call terminal 12 can first sample the voice signal after collecting the user's voice signal to obtain a sampled signal. The sampled signal can be used as a reference to determine which voice signal in the mixed signal needs to be eliminated. A target mixed signal containing the voice signal can then be obtained from the mixed signal returned by the call host. After a certain amplitude gain is applied to the sampled signal, it is superimposed on the target mixed signal to eliminate the voice signal in the target mixed signal. For example, if the voice signal in the sampled signal and the target mixed signal is actually the same signal, the sampled signal and the target mixed signal can be operated on using an inverse algorithm to eliminate the voice signal in the target mixed signal.
[0044] In some embodiments, in order to quickly and accurately obtain the target mixed signal containing the voice signal from the mixed signal returned by the call host 11, the delay generated by the mixing processing performed by the call host 11 can be determined first, and then the target mixed signal containing the voice signal can be obtained from the mixed signal returned by the call host 11 based on the delay.
[0045] For example, as shown in FIG4 , after the call terminal 12 collects the voice signal A through the microphone at time T1, it can use the analog-to-digital converter (ADC) to convert the collected analog voice signal A into a digital signal A. The digital voice signal A can then be sampled and processed to obtain a sampled signal A'. The digital voice signal A is then sent to the call host 11 via a wireless transmission unit (e.g., a DECT wireless transmission unit) for mixing processing, thereby obtaining a mixed signal including the voice signal A. Considering the time delay caused by the mixing process, which includes the time t1 required for the call terminal 12 to transmit the collected voice signal to the call host 11, the time t2 required for the call host 11 to perform the mixing process, and the time t3 required for the call host 11 to transmit the mixed signal to the call terminal 12, the time it takes for the call terminal 12 to receive the mixed signal including the voice signal A is: T2 = T1 + t1 + t2 + t3. Therefore, the call terminal 12 can receive a mixed signal (digital signal) from the call host 11, and then obtain the mixed signal received from the call host 11 at time T2 (i.e., the target mixed signal), and then perform a certain amplitude gain processing on the sampling signal A', and then perform a certain reverse operation on the sampling signal and the target mixed signal to eliminate the voice signal A in the target mixed signal with the help of the sampling signal A', and then the target voice signal after the mute processing can be converted into an analog voice signal through a digital-to-analog converter (DAC), and the analog voice signal can be played through the speaker.
[0046] Among them, the above-mentioned t1 can be determined based on the uplink time slot allocated to the call terminal, the above-mentioned t3 can be determined based on the downlink time slot allocated to the call terminal t2, and the above-mentioned t2 can be determined based on the performance of the mixing processing unit.
[0047] Of course, in actual applications, when obtaining the target mixed signal including the voice signal from the mixed signal returned from the call host 11, in addition to being based on time delay, other methods can also be used. For example, the mixed signal can be analyzed to determine whether there is a voice signal consistent with the sampling signal. If so, it means that the mixed signal includes the voice signal.
[0048] In some embodiments, the communication timeslots between the call host 11 and multiple call terminals 12 include downlink timeslots. The call host 11 broadcasts mixed audio signals to the multiple call terminals 12 via the downlink timeslots. All call terminals 12 within the same call group share a single downlink timeslot. That is, after receiving a voice signal from a call terminal 12 within a call group, the call host 11 can mix the received voice signal to generate a mixed audio signal, which is then broadcast to each call terminal 12 within that group via a single downlink timeslot. This saves timeslots and improves timeslot utilization.
[0049] Consider that within the same work scenario, different users may have varying functional requirements for call terminals 12. For example, some users speak frequently throughout their work process and require real-time communication. Other users spend the majority of their work time listening to others speak and speak less frequently. Still others simply listen to others speak and do not need to speak themselves. In the prior art, all call terminals 12 within a call system operate in full-duplex mode, which limits the number of call terminals 12 that can be connected.
[0050] To increase the number of call terminals 12 that can connect to the call system, different operating modes can be configured for each user's call terminal 12 based on their needs. For example, when joining the call system, a call terminal 12 can first send a pairing request to the call host 11. After receiving the pairing request, the call host 11 can establish a connection with the call terminal 12 and configure an operating mode for the call terminal 12. The operating mode can include any of the following: full-duplex mode, PTT mode, and simplex mode.
[0051] For example, a user's call terminal 12, which speaks frequently throughout a session and needs to be able to speak in real time, can be configured in full-duplex mode. A dedicated uplink timeslot (hereinafter referred to as a private uplink timeslot) can be allocated to the full-duplex mode call terminal 12 so that it can collect the user's voice signal at any time and send it to the call host 11 via the private uplink timeslot.
[0052] For the call terminals 12 of users who speak occasionally during the entire work process, they can be configured to PTT mode. All call terminals 12 in PTT mode can share one or more uplink time slots (hereinafter referred to as public uplink time slots). When all call terminals 12 in PTT mode want to speak, they can first apply for an idle public uplink time slot from the call host 11, and then use the public uplink time slot to send the collected user voice signal to the call host 11. Among them, for the call terminal 12 in PTT mode, when the public uplink time slot is not allocated to the call terminal 12, the call terminal 12 is in simplex mode, that is, it can only listen but not speak. After the public uplink time slot is allocated to the call terminal 12, the call terminal 12 switches to full-duplex mode, that is, it can both listen and speak.
[0053] For users whose call terminals 12 only need to listen to others' conversations during the entire conversation, they can be configured in simplex mode. A call terminal 12 in simplex mode only shares a downlink timeslot with other call terminals 12 in the same group and does not occupy uplink timeslots. In other words, a call terminal in simplex mode can only listen to others' conversations and cannot speak itself.
[0054] After configuring the operating mode of each call terminal 12, time slots can be allocated based on the operating mode of each call terminal 12. For example, a downlink time slot can be allocated to each call terminal 12 in a call group, a private uplink time slot can be allocated to a call terminal 12 in full-duplex mode, and a public uplink time slot can be allocated to a call terminal 12 in PTT mode.
[0055] In some embodiments, the operating mode corresponding to each call terminal 12 is determined based on a user's mode selection instruction. For example, during the pairing process between the call terminal 12 and the call host 11, the call host 11 may display an operating mode configuration interface through a user interface, and the user may configure the operating mode corresponding to each call terminal 11 in this configuration interface.
[0056] In some embodiments, the operating mode corresponding to the call terminal 12 can also be dynamically adjusted based on the number of call terminals 12 currently connected to the call host 11 and the number of time slots. For example, if the current number of time slots is sufficient for every call terminal 12 in the call system to operate in full-duplex mode, all call terminals 12 will be configured to operate in full-duplex mode.
[0057] In some embodiments, at least some of the call terminals 12 include call terminals in full-duplex mode. The communication time slots between the call host 11 and the multiple call terminals 12 include private uplink time slots. Each call terminal 12 in full-duplex mode occupies a private uplink time slot and sends its own collected voice signal to the call host 11 through the private uplink time slot. Since the call terminal 12 in full-duplex mode occupies an uplink time slot alone, it can send voice signals to the call host 11 at any time, that is, it can speak and listen at any time. For example, as shown in Figure 5, taking the use of DECT technology for communication between the call host and the call terminal as an example, assuming that all the call terminals 12 in the call system are a call group, the number of call terminals 12 working in full-duplex mode can be increased to 17 by adopting the solution of the present application.
[0058] Considering that even if the mixing processing logic in the call host 11 is improved so that all call terminals 12 in the same call group can share a downlink time slot, thereby freeing up more time slots for use as uplink time slots so that more call terminals 12 can be connected, the number of time slots is limited after all, and the number of call terminals 12 will still be subject to certain restrictions. Considering that in actual application scenarios, some staff members spend most of their time listening to others and rarely speak themselves, these users do not actually need to be in full-duplex mode all the time. They only need to be temporarily in full-duplex mode when they need to speak, and an uplink time slot is allocated to them. Therefore, in some embodiments, at least some of the call terminals 12 include call terminals 12 in PTT mode, and the communication time slots between the call host 11 and multiple call terminals 12 include a common uplink time slot. The call terminal 12 in PTT mode can send its own collected voice signal to the call host 11 through the common uplink time slot temporarily allocated by the call host 11. For example, as shown in FIG6 , the call terminals 12 (PU1-PU13) can operate in full-duplex mode, and the call terminals 12 (PU14-PUn) can operate in PTT mode. These call terminals 12 share four common time slots (17-20).
[0059] For example, one or more communication time slots can be reserved from the communication time slots of the call host 11 and multiple call terminals 12 as common uplink time slots. These common uplink time slots can be shared by call terminals 12 in PTT mode, and can be requested from the call host 11 before use. For call terminals 12 with less frequent conversations, they can be configured in PTT mode. This means that when a user needs to speak, a control signal can be sent to the call terminal 12, causing it to temporarily request a common uplink time slot from the call host 11 for transmitting the currently captured user's voice signal.
[0060] The number of public uplink time slots can be flexibly adjusted according to the number of call terminals 11 in PTT mode. For example, if the number of call terminals in PTT mode is relatively large, more public uplink time slots can be set, otherwise, fewer public uplink time slots can be set.
[0061] In some embodiments, the call terminal 12 may include a designated control component. When a user of the call terminal 12 in PTT mode wants to speak, the user may input a control signal through the control component. For example, the control component may be a physical button on the call terminal 12 or a virtual button on the screen of the call terminal 12. When the user wants to speak, the control component may be operated. After detecting the control signal input by the user through the designated control component, a time slot allocation request may be sent to the call host 11. After receiving the time slot allocation request, the call host 11 may allocate a public uplink time slot to the call terminal 12 in PTT mode based on the priority of the call terminal 12 in PTT mode and / or the occupancy of the public uplink time slot.
[0062] In some embodiments, the call host 11 may directly allocate a public uplink timeslot based on the occupancy of the public uplink timeslots. For example, if there is an unoccupied public uplink timeslot, the unoccupied public uplink timeslot is directly allocated to the call host; otherwise, the timeslot allocation request is rejected.
[0063] In some embodiments, given that some users have higher priority in actual application scenarios, it is necessary to prioritize these users to ensure timely speaking. Therefore, priorities can be pre-set for call terminals 12 in PTT mode, allowing higher-priority call terminals 12 to preempt public uplink timeslots already occupied by lower-priority call terminals 12. For example, upon receiving a timeslot allocation request from a call terminal 12 in PTT mode, if the call host 11 determines that an unoccupied public uplink timeslot exists, it directly allocates an unoccupied public uplink timeslot to the call terminal 12 in PTT mode. If it determines that no unoccupied public uplink timeslot exists, it then determines whether a target call terminal 12 currently occupying a public uplink timeslot exists, where the target call terminal 12 has a lower priority than the call terminal 12 in PTT mode. If so, the public timeslot occupied by the target call terminal 12 is vacated for the call terminal 12 in PTT mode. Of course, if the target call terminal does not exist, the timeslot allocation request is rejected.
[0064] In this way, for the call terminal 12 in the PTT mode, a public uplink time slot can also be allocated to it based on the speaking priority of the users using the call terminal 12, so as to ensure that these users can speak in time.
[0065] In some embodiments, the multiple call terminals 12 also include call terminals 12 in simplex mode, where the call terminals 12 in simplex mode are used to receive and play the mixed audio signal sent by the call host 11. Considering that in some scenarios, some users only need to listen to the speeches of other users in the group and do not need to speak themselves, the call terminals 12 used by such users can be configured in simplex mode. That is, no uplink time slots need to be allocated to them. They only need to receive the mixed audio signal from the call host 11 through the downlink time slot shared by the call terminals 12 in the group, achieving a listening-only function. As shown in Figure 7, call terminals 12 (PU1-PU13) can operate in full-duplex mode, and call terminals 12 (PU14-PUn) can operate in PTT mode. These call terminals 12 share four common time slots (17-20). Call terminals 12 (PUn+1-PUm) operate in simplex mode and do not need to occupy uplink time slots.
[0066] In addition, the present application also provides a call host, wherein the call host is communicatively connected to a plurality of call terminals, and the plurality of call terminals are divided into one or more call groups;
[0067] The call host is used to receive voice signals collected by at least some of the multiple call terminals, and for each call group, mix the voice signals sent by the call terminals in the call group to obtain a mixed signal, and send the mixed signal to each call terminal in the call group, so that at least some of the call terminals are also used to mute the received mixed signal to eliminate the voice signals collected by themselves, and play the mixed signal after mute processing.
[0068] In some embodiments, the communication time slots between the call host and the multiple call terminals include downlink time slots, and the call host broadcasts mixed signals to the multiple call terminals through the downlink time slots, wherein all call terminals in the same call group share one downlink time slot.
[0069] In some embodiments, at least some of the call terminals include call terminals in full-duplex mode, and the communication time slots between the call host and the multiple call terminals include private uplink time slots. Each call terminal in full-duplex mode occupies a private uplink time slot, and the call host receives the voice signal sent by the call terminal in full-duplex mode through the private uplink time slot.
[0070] In some embodiments, at least some of the call terminals include call terminals in PTT mode, and the communication time slots between the call host and the multiple call terminals include common uplink time slots. The call host is used to allocate common uplink time slots to the call terminals in PTT mode based on the priority of the call terminals in PTT mode and the occupancy of the common uplink time slots after receiving the time slot allocation request sent by the call terminals in PTT mode, and receive the voice signals sent by the call terminals in PTT mode through the common uplink time slots.
[0071] In some embodiments, the call host allocates a public uplink time slot to the call terminal in the PTT mode based on the priority of the call terminal in the PTT mode and the occupancy of the public uplink time slot, specifically for:
[0072] If there is currently an unoccupied public uplink time slot, directly allocate an unoccupied public uplink time slot to the calling terminal in the PTT mode;
[0073] If there is no unoccupied public uplink time slot, determining whether there is a target call terminal among the call terminals currently occupying the public uplink time slot, the priority of the target call terminal being lower than the priority of the call terminal in the PTT mode;
[0074] If so, the public time slot occupied by the target call terminal is vacated for the call terminal in PTT mode.
[0075] In some embodiments, the call host is also used to establish a connection with the call terminal after receiving a pairing request from the call terminal, configure a working mode for the call terminal, and allocate a communication time slot for the call terminal based on the configured working mode. The working mode includes any one of the following: full-duplex mode, PTT mode, and simplex mode.
[0076] In some embodiments, the working mode corresponding to each call terminal is determined based on the user's mode selection instruction.
[0077] The specific details of how the call host implements various functions can be found in the description of the above embodiments and will not be repeated here.
[0078] In addition, an embodiment of the present application further provides a call terminal, wherein the call terminal is communicatively connected to a call host;
[0079] The call terminal is used to collect the user's voice signal and send the voice signal to the call host so that the call host mixes the received voice signal to obtain a mixed signal; and, receives the mixed signal sent by the call host, performs silencing on the received mixed signal to eliminate the voice signal collected by itself, and plays the mixed signal after silencing.
[0080] In some embodiments, the call terminal is configured to perform silencing processing on the received mixed signal to eliminate the voice signal collected by itself, specifically for:
[0081] Sampling and processing the voice signal collected by itself to obtain a sampled signal;
[0082] Obtaining a target mixed signal containing the voice signal from the mixed signal returned by the call host;
[0083] After performing gain processing on the sampled signal, the sampled signal is added to the target mixed signal to eliminate the speech signal in the target mixed signal.
[0084] In some embodiments, when the call terminal is used to obtain the target mixed signal containing the voice signal from the mixed signal returned by the call host, it is specifically used to:
[0085] Determining the delay caused by the mixing process performed by the call host;
[0086] A target mixed signal including the voice signal is obtained from the mixed signal returned by the call host based on the time delay.
[0087] In some embodiments, if the call terminal is in full-duplex mode, the call terminal occupies a private uplink time slot and sends the voice signal collected by itself to the call host through the private uplink time slot.
[0088] In some embodiments, if the call terminal is in PTT mode, the call terminal shares one or more common uplink time slots with other call terminals in PTT mode;
[0089] The call terminal is used to send a time slot allocation request to the call host after detecting a control signal input by the user through a designated control component, so that the call host, after receiving the time slot allocation request, allocates a public uplink time slot to the call terminal based on the priority of the call terminal and the occupancy of the public uplink time slot.
[0090] In some embodiments, if the call terminal is in simplex mode, the call terminal is used to receive the mixed signal broadcast by the call host and play the mixed signal.
[0091] Among them, the specific details of how the call terminal implements each function can be referred to the description in the above embodiment and will not be repeated here.
[0092] The various technical features in the above embodiments can be arbitrarily combined as long as there is no conflict or contradiction between the combinations of features. However, due to space limitations, they are not described one by one. Therefore, the arbitrary combination of the various technical features in the above embodiments also falls within the scope of disclosure of this specification.
[0093] Those skilled in the art will readily recognize other implementations of the embodiments of the present invention after considering the specification and practicing the instructions disclosed herein. The embodiments of the present invention are intended to cover any variations, uses, or adaptations of the embodiments of the present invention that follow the general principles of the embodiments of the present invention and include common knowledge or customary techniques in the art not disclosed in the embodiments of the present invention. The description and examples are to be considered as exemplary only, and the true scope and spirit of the embodiments of the present invention are indicated by the following claims.
[0094] It should be understood that the embodiments of the present invention are not limited to the precise structures 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 invention is limited only by the appended claims.
[0095] The above description is only a preferred embodiment of the embodiments of this specification and is not intended to limit the embodiments of this specification. Any modifications, equivalent replacements, improvements, etc. made within the spirit and principles of the embodiments of this specification should be included in the scope of protection of the embodiments of this specification.
Claims
1. A communication system, characterized in that: The call system includes a call host and a plurality of call terminals, the call host is wirelessly connected to the plurality of call terminals respectively, and the plurality of call terminals are divided into one or more call groups; At least some of the multiple call terminals are used to collect voice signals of users and send the voice signals to the call host; The call host is used to mix the voice signals sent by the call terminals in each call group to obtain a mixed signal, and send the mixed signal to each call terminal in the call group; At least some of the call terminals are also used to mute the received mixed signal to eliminate the voice signal collected by themselves, and play the mixed signal after the mute processing.
2. The communication system according to claim 1, characterized in that: When at least some of the call terminals are used to perform noise elimination processing on the received mixed signal to eliminate the voice signal collected by themselves, they are specifically used to: Sampling and processing the voice signal collected by itself to obtain a sampled signal; Acquire a target mixed signal including the voice signal from the mixed signal returned by the call host; After gain processing is performed on the sampled signal, the sampled signal is added to the target mixed signal to eliminate the speech signal in the target mixed signal.
3. The communication system according to claim 2, characterized in that: When the at least some call terminals are used to obtain the target mixed signal containing the voice signal from the mixed signal returned by the call host, it is specifically used to: Determine the delay caused by the mixing process performed by the call host; A target mixed audio signal including the voice signal is obtained from the mixed audio signal returned by the call host based on the time delay.
4. The communication system according to any one of claims 1 to 3, characterized in that: The communication time slots between the call host and the multiple call terminals include downlink time slots, and the call host broadcasts mixed signals to the multiple call terminals through the downlink time slots, wherein all call terminals in the same call group share one downlink time slot.
5. The communication system according to any one of claims 1 to 4, characterized in that: At least some of the call terminals include call terminals in full-duplex mode, and the communication time slots between the call host and the multiple call terminals include private uplink time slots. Each call terminal in full-duplex mode occupies a private uplink time slot and sends its own collected voice signals to the call host through the private uplink time slot.
6. The communication system according to any one of claims 1 to 5, characterized in that: At least some of the call terminals include call terminals in PTT mode, and the communication time slots between the call host and the multiple call terminals include common uplink time slots. The call terminals in PTT mode are used to send their own collected voice signals to the call host through the common uplink time slots temporarily allocated by the call host.
7. The communication system according to claim 6, characterized in that: The number of the common uplink time slots is positively correlated with the number of call terminals in the PTT mode.
8. The communication system according to claim 6 or 7, characterized in that: The call terminal in PTT mode is used to send a time slot allocation request to the call host after detecting a control signal input by the user through a designated control component; the call host is used to allocate a public uplink time slot to the call terminal in PTT mode based on the priority of the call terminal in PTT mode and the occupancy of the public uplink time slot after receiving the time slot allocation request.
9. The communication system according to claim 8, characterized in that: The call host allocates a common uplink time slot to the call terminal in the PTT mode based on the priority of the call terminal in the PTT mode and the occupancy of the common uplink time slot, specifically for: If there is currently an unoccupied public uplink time slot, directly allocating an unoccupied public uplink time slot to the call terminal in the PTT mode; If there is no unoccupied common uplink time slot currently, it is determined that the common uplink time slot currently occupied whether there is a target call terminal among the call terminals in the row time slot, and the priority of the target call terminal is lower than the priority of the call terminal in the PTT mode; If so, the public time slot occupied by the target call terminal will be vacated for the call terminal in PTT mode.
10. The communication system according to any one of claims 1 to 9, characterized in that: The multiple call terminals also include a call terminal in simplex mode, and the call terminal in simplex mode is used to receive a mixed signal sent by the call host and play the mixed signal.
11. The communication system according to any one of claims 1 to 10, characterized in that: The call host is also used to establish a connection with the call terminal after receiving a pairing request from the call terminal, configure a working mode for the call terminal, and allocate a communication time slot for the call terminal based on the configured working mode. The working mode includes any one of the following: full-duplex mode, PTT mode, and simplex mode.
12. The communication system according to claim 11, characterized in that: The working mode corresponding to each call terminal is determined based on the user's mode selection instruction.
13. A call host, characterized in that: The call host is respectively connected to a plurality of call terminals for communication, and the plurality of call terminals are divided into one or more call groups; The call host is used to receive voice signals collected by at least some of the multiple call terminals, and for each call group, mix the voice signals sent by the call terminals in the call group to obtain a mixed signal, and send the mixed signal to each call terminal in the call group, so that at least some of the call terminals can mute the received mixed signals to eliminate the voice signals collected by themselves, and play the mixed signals after mute processing.
14. The communication host according to claim 13, characterized in that: The communication time slots between the call host and the multiple call terminals include downlink time slots, and the call host broadcasts mixed signals to the multiple call terminals through the downlink time slots, wherein all call terminals in the same call group share one downlink time slot.
15. The communication host according to claim 13, characterized in that: At least some of the call terminals include call terminals in full-duplex mode, and the communication time slots between the call host and the multiple call terminals include private uplink time slots. Each call terminal in full-duplex mode occupies a private uplink time slot, and the call host receives the voice signal sent by the call terminal in full-duplex mode through the private uplink time slot.
16. The communication host according to claim 13, characterized in that: At least some of the call terminals include call terminals in PTT mode, and the communication time slots between the call host and the multiple call terminals include common uplink time slots. The call host is used to allocate common uplink time slots to the call terminals in PTT mode based on the priority of the call terminals in PTT mode and the occupancy of the common uplink time slots after receiving the time slot allocation request sent by the call terminals in PTT mode, and receive the voice signals sent by the call terminals in PTT mode through the common uplink time slots.
17. The communication host according to claim 16, characterized in that: The call host allocates a common uplink time slot to the call terminal in the PTT mode based on the priority of the call terminal in the PTT mode and the occupancy of the common uplink time slot, specifically for: If there is currently an unoccupied public uplink time slot, directly allocating an unoccupied public uplink time slot to the call terminal in the PTT mode; If there is no unoccupied public uplink time slot currently, determining whether there is a target call terminal among the call terminals currently occupying the public uplink time slot, the priority of the target call terminal being lower than the priority of the call terminal in the PTT mode; If so, the public time slot occupied by the target call terminal will be vacated for the call terminal in PTT mode.
18. The communication host according to claim 13, characterized in that: The call host is also used to establish a connection with the call terminal after receiving a pairing request from the call terminal, configure a working mode for the call terminal, and allocate a communication time slot for the call terminal based on the configured working mode. The working mode includes any one of the following: full-duplex mode, PTT mode, and simplex mode.
19. The communication system according to claim 18, characterized in that: The working mode corresponding to each call terminal is determined based on the user's mode selection instruction.
20. A call terminal, characterized in that: The call terminal is communicatively connected with the call host; The call terminal is used to collect the user's voice signal and send the voice signal to the call host, so that the call host performs mixing processing on the received voice signal to obtain a mixed signal; And, receiving the mixed signal sent by the call host, performing silencing processing on the received mixed signal to eliminate the voice signal collected by itself, and playing the mixed signal after silencing processing.
21. The call terminal according to claim 20, characterized in that: The calling terminal is used to perform silencing processing on the received mixed signal to eliminate the voice signal collected by itself, specifically for: Sampling and processing the voice signal collected by itself to obtain a sampled signal; Acquire a target mixed signal including the voice signal from the mixed signal returned by the call host; After gain processing is performed on the sampled signal, the sampled signal is added to the target mixed signal to eliminate the speech signal in the target mixed signal.
22. The call terminal according to claim 21, characterized in that: When the call terminal is used to obtain the target mixed signal containing the voice signal from the mixed signal returned by the call host, it is specifically used to: Determine the delay caused by the mixing process performed by the call host; A target mixed audio signal including the voice signal is obtained from the mixed audio signal returned by the call host based on the time delay.
23. The call terminal according to claim 21, characterized in that: If the call terminal is in full-duplex mode, the call terminal occupies a private uplink time slot and sends the voice signal collected by itself to the call host through the private uplink time slot.
24. The call terminal according to claim 21, characterized in that: If the call terminal is in PTT mode, the call terminal shares one or more common uplink time slots with other call terminals in PTT mode; The call terminal is used to send a time slot allocation request to the call host after detecting a control signal input by the user through a designated control component, so that the call host receives the time slot After the allocation request, a public uplink time slot is allocated to the call terminal based on the priority of the call terminal and the occupancy of the public uplink time slot.
25. The call terminal according to claim 21, characterized in that: If the call terminal is in simplex mode, the call terminal is used to receive the mixed signal broadcast by the call host and play the mixed signal.
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