WO2018227854A1 - Speech talkback method and apparatus, and mobile terminal - Google Patents

Speech talkback method and apparatus, and mobile terminal Download PDF

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Publication number
WO2018227854A1
WO2018227854A1 PCT/CN2017/109187 CN2017109187W WO2018227854A1 WO 2018227854 A1 WO2018227854 A1 WO 2018227854A1 CN 2017109187 W CN2017109187 W CN 2017109187W WO 2018227854 A1 WO2018227854 A1 WO 2018227854A1
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WO
WIPO (PCT)
Prior art keywords
voice
algorithm
file
server
speech
Prior art date
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PCT/CN2017/109187
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French (fr)
Chinese (zh)
Inventor
张国滔
郑勇
魏科文
卫特超
郑培艺
Original Assignee
深圳市沃特沃德股份有限公司
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Publication of WO2018227854A1 publication Critical patent/WO2018227854A1/en

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Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/087Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters using mixed excitation models, e.g. MELP, MBE, split band LPC or HVXC
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04BTRANSMISSION
    • H04B7/00Radio transmission systems, i.e. using radiation field
    • H04B7/14Relay systems
    • H04B7/15Active relay systems
    • H04B7/185Space-based or airborne stations; Stations for satellite systems
    • H04B7/1851Systems using a satellite or space-based relay
    • H04B7/18513Transmission in a satellite or space-based system
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04QSELECTING
    • H04Q5/00Selecting arrangements wherein two or more subscriber stations are connected by the same line to the exchange
    • H04Q5/24Selecting arrangements wherein two or more subscriber stations are connected by the same line to the exchange for two-party-line systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W84/00Network topologies
    • H04W84/02Hierarchically pre-organised networks, e.g. paging networks, cellular networks, WLAN [Wireless Local Area Network] or WLL [Wireless Local Loop]
    • H04W84/04Large scale networks; Deep hierarchical networks
    • H04W84/06Airborne or Satellite Networks

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a voice intercom method, apparatus, and mobile terminal.
  • satellite mobile communications The use of geostationary orbit satellites or medium and low-orbit satellites as relay stations to achieve regional or global mobile communications is known as satellite mobile communications. It generally consists of three parts: a communication satellite, consisting of one or more satellites; a ground station, including a system control center and several gateway stations (ie, transit stations that connect the public switched telephone network to mobile users); mobile users Communication terminals, including onboard, shipboard, airborne terminals and handsets. The user can move freely within the coverage of the satellite beam, and the satellite transmits signals to maintain communication with terrestrial communication systems and dedicated system users or other mobile users.
  • a communication satellite consisting of one or more satellites
  • a ground station including a system control center and several gateway stations (ie, transit stations that connect the public switched telephone network to mobile users)
  • mobile users Communication terminals, including onboard, shipboard, airborne terminals and handsets. The user can move freely within the coverage of the satellite beam, and the satellite transmits signals to maintain communication with terrestrial communication systems and dedicated system users or other mobile users.
  • satellite mobile communication Compared with other communication methods, satellite mobile communication has the advantages of large coverage area, long communication distance, flexible communication and maneuverability, and stable and reliable lines. Therefore, satellite mobile communication has become an important development direction of communication services.
  • the satellite mobile communication network Like the public land mobile communication network, the satellite mobile communication network also supports the function of the TCP/IP link to access the Internet, so that the mobile terminal can communicate via the satellite mobile communication network.
  • the bandwidth of the satellite mobile communication network is narrow, it is impossible to implement an actual voice intercom using the instant communication application, thereby affecting the user experience.
  • a primary object of the present invention is to provide a voice intercom method, apparatus, and mobile terminal, which are directed to solving a technical problem that a mobile terminal based on satellite mobile communication cannot implement a real voice intercom using an instant communication application.
  • an embodiment of the present invention provides a voice intercom method, where the method includes the following steps. [0007] maintaining a long connection with the server through the satellite mobile communication network;
  • the method further includes:
  • the step of obtaining, by the server, the second voice file sent by the peer end includes:
  • the step of outputting the second voice file includes:
  • the second speech file is decoded using a low rate speech decoding algorithm and played.
  • the low-rate speech coding algorithm is an adaptive multi-rate AMR algorithm, a hybrid excitation linear predictive coding MELP algorithm, a code excitation linear predictive coding CELP algorithm, a sine transform coding STC algorithm, and a frequency domain interpolation coding TFI.
  • Algorithm pitch synchronization excitation linear prediction coding PSELP algorithm, multi-band excitation coding MBE algorithm or waveform interpolation coding WI algorithm.
  • the step of compressing the collected voice information by using a low-rate speech compression coding algorithm and generating a voice file includes:
  • the collected voice information is compression-encoded by using the AMR algorithm to reduce the bit rate of the voice information to a preset value, and generate a voice file in an AMR format.
  • the preset value is 6.6 kb/s.
  • the step of compressing the collected voice information by using a low-rate voice compression coding algorithm and generating a voice file includes: [0024]
  • the collected voice information is compression-encoded by using the MELP algorithm to reduce the code rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format.
  • the step of sending the voice file to the server includes:
  • the voice file is transmitted to the server in a packet using a TCP/IP protocol.
  • the embodiment of the present invention provides a voice intercom device, and the device includes:
  • connection module configured to maintain a long connection with the server through the satellite mobile communication network
  • a processing module configured to compress the collected voice information by using a low-rate speech coding algorithm and generate a first voice file
  • a sending module configured to send the first voice file to the server, so that the server sends the first voice file to the peer end.
  • the device further includes:
  • an obtaining module configured to acquire, from the server, a second voice file sent by the peer end
  • an output module configured to output the second voice file.
  • the acquiring module includes:
  • a receiving unit configured to receive a download address of the second voice file sent by the server
  • a downloading unit configured to download the second voice file according to the download address.
  • the output module includes:
  • a determining unit configured to determine whether the second voice file is a low-rate voice encoded file
  • a playing unit configured to: when the file is a low-rate speech encoded file, play the second voice file by using a low-rate speech decoding algorithm.
  • the processing module is used to convert the low-rate speech compression coding algorithm into AMR algorithm
  • the processing module is used to determine whether the low-rate voice compression coding algorithm is a MELP algorithm.
  • the sending module is configured to: packet transmit the voice file to the server by using a TCP/IP protocol.
  • the present invention also proposes a mobile terminal, the mobile terminal comprising a memory, a processor and at least one application stored in the memory and configured to be executed by the processor, characterized in that The application is configured to perform the aforementioned voice intercom method.
  • a voice intercom method provided by an embodiment of the present invention maintains a long connection with a server through a satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing voice information.
  • the code rate reduces the capacity of the transmitted voice file, saves the bandwidth resources of the satellite mobile communication network, and realizes the low-end, short-talk, intercom, and solves the problem that the mobile terminal based on satellite mobile communication in the prior art cannot be utilized.
  • the communication application realizes the technical problem of the voice intercom and improves the user experience.
  • FIG. 1 is a flow chart of a first embodiment of a voice intercom method of the present invention
  • FIG. 2 is a flow chart of a second embodiment of a voice intercom method of the present invention.
  • FIG. 3 is a block diagram showing a first embodiment of a voice intercom apparatus of the present invention.
  • FIG. 4 is a block diagram showing a second embodiment of a voice intercom apparatus of the present invention.
  • FIG. 5 is a block diagram of an acquisition module of the voice intercom device of FIG. 4;
  • FIG. 6 is a block diagram of an output module of the voice intercom device of FIG. 4.
  • terminal and terminal device used herein include both a device of a wireless signal receiver, a device having only a wireless signal receiver without a transmitting capability, and a receiving and receiving device.
  • Such a device may comprise: a cellular or other communication device having a single line display or a multi-line display or a cellular or other communication device without a multi-line display; PCS (Persona 1 Communications Service), which may combine voice, Data processing, fax and/or data communication capabilities; PDA (Personal Digital Assistant), which can include radio frequency receivers, pagers, Internet/Intranet access, web browsers, notepads, calendars and/or GPS ( Global Positioning System, Receiver; Conventional laptop and/or palmtop computer or other device having a conventional laptop and/or palmtop computer or other device that includes and/or includes a radio frequency receiver.
  • PCS Personala 1 Communications Service
  • PDA Personal Digital Assistant
  • terminal may be portable, transportable, installed in a vehicle (aviation, sea and/or land), or adapted and/or configured to operate locally, and/or Run in any other location on the Earth and/or space in a distributed fashion.
  • the "terminal” and “terminal device” used herein may also be a communication terminal, an internet terminal, a music/video playback terminal, For example, it may be a PDA, a MID (Mobile Internet Device), and/or a mobile phone with music/video playback function, or a smart TV, a set top box, or the like.
  • the server used herein includes, but is not limited to, a computer, a network host, a single network server, a plurality of network server sets, or a cloud composed of a plurality of servers.
  • the cloud consists of a large number of computers or network servers based on Cloud Computing, which is a kind of distributed computing, a super virtual computer composed of a group of loosely coupled computers.
  • communication may be implemented by any communication means between the server, the terminal device and the WNS server, including but not limited to, mobile communication based on 3GPP, LTE, WIMAX, and computer network communication based on TCP/IP and UDP protocols. And short-range wireless transmission based on Bluetooth and infrared transmission standards.
  • a first embodiment of a voice intercom method of the present invention is proposed.
  • the method includes the following steps: [0059] Sl l, maintaining a long connection with a server through a satellite mobile communication network.
  • step S11 after the mobile terminal establishes a connection with the server through the satellite mobile communication network, the mobile terminal maintains a long connection with the server with a certain heartbeat period, that is, the mobile terminal sends a heartbeat packet to the server every other heartbeat period, thereby The connection path between the two is maintained, so that the subsequent voice packets are delayed and transmitted.
  • the mobile terminal may maintain a long connection with the server in a preset heartbeat period.
  • the mobile terminal may adaptively adjust the heartbeat period according to the reference heartbeat period and the signal quality of the satellite mobile communication network, acquire an adaptive heartbeat period with a large period and maintain a stable connection, and adopt an adaptive heartbeat The cycle maintains a long connection with the server.
  • the reference heartbeat period may be a preset heartbeat period, a heartbeat period used by the last connection, a heartbeat period used by other mobile communication networks (such as the public land mobile communication network), and the like.
  • the mobile terminal first performs a long connection test with a reference heartbeat period.
  • the length of the reference heartbeat period is increased to perform a long connection test to obtain an adaptive heartbeat period capable of maintaining a long connection, for example: increasing the length of the reference based on the reference heartbeat period Long connection test until the long connection cannot be maintained;
  • the heartbeat period of the previous test is selected as the adaptive heartbeat period.
  • the long connection test is performed on the basis of the reference heartbeat period to obtain a long connection test.
  • the adaptive heartbeat period for example: based on the reference heartbeat period, successively reduces the length of the long connection test until the long connection can be maintained; when the long connection is maintained, the heartbeat period of the test is selected as the adaptive heartbeat period. Finally, the mobile terminal maintains a long connection with the server in an adaptive heartbeat cycle.
  • the mobile terminal may be a satellite mobile communication terminal supporting only satellite mobile communication, or a fusion terminal supporting both satellite mobile communication and public land mobile communication for satellite mobile communication and public land mobile communication.
  • S12. Process the collected voice information by using a low-rate speech coding algorithm and generate a first voice file.
  • the mobile terminal can perform voice intercom with other terminals through a short communication service such as WeChat, E-mail, and QQ, and the other terminals are the opposite ends of the mobile terminal.
  • a short communication service such as WeChat, E-mail, and QQ
  • the mobile terminal sends a voice file to the opposite end
  • the mobile terminal is the transmitting end, and the opposite end is the receiving end
  • the mobile terminal receives the voice file sent by the opposite end
  • the mobile terminal is the receiving end, and the opposite end is the transmitting end.
  • the voice information is collected through the microphone, and the collected voice information is processed by using a low-rate voice encoding algorithm to generate a first voice file.
  • the mobile terminal collects voice information
  • the application processor collects voice information by using an 8-bit ADC and a sampling frequency of 8 k, and digitally records the collected voice information.
  • the code rate of the collected voice information is 64 kb/s.
  • the low rate speech coding algorithm may be an adaptive multi-rate (AMR) algorithm, a mixed excitation linear predictive coding (MELP) algorithm, a code excited linear predictive coding (CELP) algorithm, a sinusoidal transform coding (STC) algorithm, a chirp frequency domain.
  • AMR adaptive multi-rate
  • MELP mixed excitation linear predictive coding
  • CELP code excited linear predictive coding
  • STC sinusoidal transform coding
  • TFI Interpolation Coding
  • PSELP Pitch Synchronous Linear Predictive Coding
  • MBE Multi-Band Excitation Coding
  • WI Waveform Interpolation Coding
  • the mobile terminal uses the AMR algorithm to compress and encode the collected voice information, so as to reduce the code rate of the voice information to a preset value, and generate a voice file in the AMR format.
  • the AMR can use nine codes from 6.6 kb/s to 23.85 kb/s, and the preset value is preferably a minimum code rate of 6.6 kb/s.
  • the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized.
  • the mobile terminal uses the MELP algorithm to compress and encode the collected voice information to reduce the bit rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format.
  • the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized.
  • the satellite communication modem of the mobile terminal establishes communication with the server through a socket, and the mobile terminal preferably adopts a Transmission Control Protocol/Internet Protocol (T CP/IP) protocol.
  • T CP/IP Transmission Control Protocol/Internet Protocol
  • the package transmits voice files to the server. That is, the mobile terminal divides the first voice file into a plurality of voice packets, and sequentially transmits the plurality of voice packets to the server in sequence.
  • the server After receiving the voice packets, stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the first voice file.
  • Each voice packet is a TCP/IP protocol packet, and the components of the TCP/IP protocol packet are as follows:
  • the mobile terminal and the server may agree on the definition of the packet header (such as setting different identifiers), and the server parses the packet header of the TCP/IP protocol packet to distinguish whether the network transmitting the voice file is a satellite mobile communication network or a public land mobile.
  • the communication network that is, distinguishes whether the voice file sent by the sender is a low rate voice code file or a normal voice code file.
  • the server may adopt a software architecture that supports concurrent access of multiple clients, such as MINA, Erlang, etc.
  • the specific process on the server side is: creating a server-side object to generate a listening thread, starting port listening, and accepting a client connection request, when a client connection is up, creating a client object to generate a new thread.
  • Send data to the client create a data stream transfer object, start data snooping, and when receiving data, determine the data length. When the data length is 0 ⁇ , it is determined that the connection has been broken, and the client object and the useless thread are deleted; when the data length is not 0 ⁇ , the data is processed.
  • the server may send the voice file to the receiving end in the following two ways: One is to send the download address of the first voice file to the receiving end, so that the receiving end directly downloads according to the download address.
  • the first voice file the other is to use the TCP/IP protocol to sub-packet the first voice file to the next Received.
  • the downloading method is preferred, and the delay can be reduced. If the receiving end accesses the public land mobile communication network, both methods are available.
  • the voice intercom method of the embodiment of the present invention maintains a long connection with the server through the satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing the bit rate of the voice information.
  • the capacity of the transmitted voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized, and the mobile terminal based on the satellite mobile communication in the prior art cannot be solved by using the instant communication application.
  • the technical problems of the actual voice intercom have improved the user experience.
  • step S11 when the mobile terminal is the receiving end, step S11 further includes:
  • the server preferably sends the download address of the second voice file to the mobile terminal, the mobile terminal receives the download address sent by the server, and downloads the second voice file according to the download address.
  • the server may also use the TCP/IP protocol to packetize and transmit the second voice file to the mobile terminal, that is, the server divides the second voice file into multiple voice packets, and sequentially sequentially multiple voice packets. Send to the mobile terminal. After receiving the voice packets, the mobile terminal stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the second voice file.
  • the mobile terminal after receiving the second voice file, the mobile terminal first determines whether the second voice file is a low-rate voice code file; when the voice code file is a low-rate voice code file, the mobile terminal uses the low-rate voice decoder.
  • the low-rate speech decoding algorithm decodes the second speech file and plays it; when it is a normal speech-encoded file, the second speech file is decoded by the wide-band speech decoder and then played.
  • the mobile terminal may determine whether it is a low-rate voice encoded file by using the identifier information of the second voice file, and the identifier information may be set in a packet header of the voice packet of the second voice file.
  • the second voice file is determined to be a low-rate voice encoded file; and when the identifier information of the second voice file is the second identifier, determining the second The voice file is a normal voice encoded file.
  • the identification information of the second voice file is the first identifier
  • determining that the second voice file is a low-rate voice encoded file when the identifier information of the second voice file is empty (ie, no identifier), It is determined that the second voice file is a normal voice encoded file. Or, vice versa.
  • the voice intercom method of the embodiment obtains the second voice file by using the download mode, which reduces the delay of the voice intercom and improves the user experience.
  • a voice intercom is performed with the mobile terminal accessing the satellite mobile communication network.
  • a mobile terminal accessing a satellite mobile communication network can perform voice intercommunication with other mobile terminals accessing the satellite mobile communication network, and can also access other mobiles of the public land mobile communication network.
  • the terminal performs voice intercom.
  • a first embodiment of a voice intercom apparatus of the present invention is proposed, and the apparatus is applied to a mobile terminal.
  • the device can also be applied to other terminal devices, and the device includes a connection module 10, a processing module 20, and a sending module 30, where:
  • Connection module 10 for maintaining a long connection with the server via the satellite mobile communication network.
  • connection module 10 after the connection module 10 establishes a connection with the server through the satellite mobile communication network, the connection module 10 maintains a long connection with the server with a certain heartbeat period, that is, the connection module 10 sends a heartbeat packet to the server every other heartbeat period. In order to maintain the connection path between the two, the low-speed transmission and subsequent transmission of the subsequent voice packets are realized.
  • connection module 10 may maintain a long connection with the server in a preset heartbeat period.
  • the connection module 10 can adaptively adjust the heartbeat period according to the reference heartbeat period and the signal quality of the satellite mobile communication network, and acquire an adaptive heartbeat period with a large period and maintaining a stable connection, and adaptively The heartbeat cycle stays connected to the server.
  • the reference heartbeat period may be a preset heartbeat period, a heartbeat period used by the last connection, a heartbeat period used by other mobile communication networks (such as a public land mobile communication network), and the like.
  • the connection module 10 first performs a long connection test with a reference heartbeat cycle.
  • the length of the reference heartbeat period is increased to perform a long connection test.
  • An adaptive heartbeat period capable of maintaining a long connection for example: increasing the length of the long connection test on the basis of the reference heartbeat period until the long connection cannot be maintained; when the long connection cannot be maintained, the heartbeat period of the previous test is selected as Adaptive heartbeat cycle.
  • the long connection test is performed on the basis of the reference heartbeat period to obtain an adaptive heartbeat period capable of maintaining a long connection, for example, the length of the reference is continuously reduced based on the reference heartbeat period. Long connection test until long connection can be maintained; When long connection is maintained, the heartbeat period of this test is selected as the adaptive heartbeat period.
  • the connection module 10 maintains a long connection with the server in an adaptive heartbeat cycle.
  • the mobile terminal may be a satellite mobile communication terminal supporting only satellite mobile communication, or a fusion terminal supporting both satellite mobile communication and public land mobile communication for satellite mobile communication and public land mobile communication.
  • the processing module 20 is configured to perform compression processing on the collected voice information by using a low-rate voice coding algorithm and generate a first voice file.
  • the voice intercom device can perform voice intercom with other terminals through a short communication service such as WeChat, E-mail, and QQ, and the other terminals are the opposite ends of the mobile terminal.
  • a short communication service such as WeChat, E-mail, and QQ
  • the other terminals are the opposite ends of the mobile terminal.
  • the processing module 20 collects the voice information through the microphone, and processes the collected voice information by using a low-rate voice encoding algorithm to generate a first voice file.
  • the processing module 20 collects voice information, and uses the 8-bit ADC and the 8K sampling frequency to collect voice information, and digitally records the collected voice information.
  • the code rate of the collected voice information is 64 kb/s.
  • the low-rate speech coding algorithm may be an adaptive multi-rate (AMR) algorithm, a mixed excitation linear predictive coding (MELP) algorithm, a code excited linear predictive coding (CELP) algorithm, a sinusoidal transform coding (STC) algorithm, and a frequency domain.
  • AMR adaptive multi-rate
  • MELP mixed excitation linear predictive coding
  • CELP code excited linear predictive coding
  • STC sinusoidal transform coding
  • TFI Interpolation Coding
  • PSELP Pitch Synchronous Linear Predictive Coding
  • MBE Multi-Band Excitation Coding
  • WI Waveform Interpolation Coding
  • the processing module 20 uses the AMR algorithm to compress the collected voice information. Reduce encoding to reduce the bit rate of voice information to a preset value, and generate a voice file in AMR format.
  • the AMR can use nine codes from 6.6 kb/s to 23.85 kb/s, and the preset value is preferably a minimum code rate of 6.6 kb/s. Thereby, the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized.
  • the processing module 20 uses the MELP algorithm to compress and encode the collected voice information to reduce the bit rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format. Therefore, the bit rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-end, short-talking intercom is realized.
  • the sending module 30 is configured to send the first voice file to the server, so that the server sends the first voice file to the peer end.
  • the sending module 30 preferably uses a TCP/IP protocol to packetize and transmit voice files to the server.
  • the transmitting module 30 divides the first voice file into a plurality of voice packets, and sequentially transmits the plurality of voice packets to the server in sequence.
  • the server After receiving the voice packets, the server stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, the first voice file is restored.
  • Each voice packet is a TCP/IP protocol packet, and the components of the TCP/IP protocol packet are as follows:
  • the sending module 30 and the server may agree on the definition of the packet header (such as setting different identifiers), and the server parses the packet header of the TCP/IP protocol packet to distinguish whether the network transmitting the voice file is a satellite mobile communication network or a public land.
  • the mobile communication network that is, distinguishes whether the voice file sent by the sender is a low rate voice code file or a normal voice code file.
  • the voice intercom device in the embodiment of the present invention maintains a long connection with the server through the satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing the bit rate of the voice information.
  • the capacity of the transmitted voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized, and the mobile terminal based on the satellite mobile communication in the prior art cannot be solved by using the instant communication application.
  • the technical problems of the actual voice intercom have improved the user experience.
  • the apparatus further includes The obtaining module 40 is configured to acquire the second voice file sent by the peer end, and the output module 50 is configured to output the second voice file.
  • the server preferably sends the download address of the second voice file to the mobile terminal.
  • the obtaining module 40 includes a receiving unit 41 and a downloading unit 42.
  • the receiving unit 41 is configured to receive a download address of the second voice file sent by the server, and the downloading unit 42 is configured to download the second voice according to the download address. file.
  • the server may also use the TCP/IP protocol to packetize and transmit the second voice file to the mobile terminal, that is, the server divides the second voice file into multiple voice packets, and sequentially sequentially multiple voice packets.
  • the obtaining module 40 receives the plurality of voice packets, and stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the second voice file.
  • the output module 50 includes a determining unit 51 and a playing unit 52, wherein: the determining unit 51 is configured to determine whether the second voice file is a low-rate voice encoded file; and the playing unit 52 is configured to use the second voice.
  • the file is a low-rate speech encoded file, and the second speech file is decoded by a low-rate speech decoder using a low-rate speech decoding algorithm; when the second speech file is a normal speech-encoded file, the broadband speech decoder is used.
  • the second voice file is decoded and played.
  • the determining unit 51 may determine whether it is a low-rate voice encoding file by using the identification information of the second voice file, and the identifier information may be set in the packet header of the voice packet of the second voice file.
  • the determining unit 51 determines that the second voice file is a low-rate voice encoded file; and when the identifier information of the second voice file is the second identifier, determining Unit 51 determines that the second voice file is a normal voice encoded file.
  • the determining unit 51 determines that the second voice file is a low-rate voice encoded file; when the identifier information of the second voice file is empty (ie, there is no identifier) ⁇ , the determining unit 51 determines that the second voice file is a normal voice encoded file. Or, vice versa.
  • the voice intercom device of the embodiment obtains the second voice file by using the download mode, which reduces the delay of the voice intercom and improves the user experience.
  • a voice intercom is performed with the mobile terminal accessing the satellite mobile communication network.
  • the present invention also proposes a mobile terminal, where the mobile terminal includes a memory, a processor, and at least one An application stored in the memory and configured to be executed by the processor, the application being configured to execute a voice intercom method.
  • the voice intercom method includes the following steps: maintaining a long connection with a server through a satellite mobile communication network; compressing the collected voice information by using a low rate speech coding algorithm to generate a first voice file; and transmitting the first message to the server a voice file, such that the server sends the first voice file to the receiving end.
  • the voice intercom method described in this embodiment is the voice intercom method involved in the foregoing embodiment of the present invention, and details are not described herein again.
  • the present invention includes apparatus related to performing one or more of the operations described herein.
  • These devices may be specially designed and manufactured for the required purposes, or may also include known devices in a general purpose computer.
  • These devices have computer programs stored therein that are selectively activated or reconfigured.
  • Such computer programs may be stored in a device (eg, computer) readable medium or in any type of medium suitable for storing electronic instructions and respectively coupled to a bus, including but not limited to any Types of disks (including floppy disks, hard disks, CDs, CD-ROMs, and magneto-optical disks), ROM (Read-Only Memory), RAM (Random Access Memory), EPROM (Erasable Programmable Read-Only)
  • a readable medium includes any medium that is stored or transmitted by a device (e.g., a computer) in a readable form.
  • each block of the block diagrams and/or block diagrams and/or flow diagrams can be implemented by computer program instructions, as well as in the block diagrams and/or block diagrams and/or flow diagrams. The combination of boxes.
  • these computer program instructions can be implemented by a general purpose computer, a professional computer, or a processor of other programmable data processing methods, such that the processor is executed by a computer or other programmable data processing method.
  • the block diagrams and/or block diagrams of the invention and/or the schemes specified in the blocks or blocks of the flow diagram are invented.

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Abstract

Disclosed are a speech talkback method and apparatus, and a mobile terminal. The method comprises the following steps: maintaining a long connection with a server by means of a satellite mobile communication network; carrying out compression processing on collected speech information by using a low-bit-rate speech encoding algorithm, and generating a first speech file; and sending the first speech file to the server so as to enable the server to send the first speech file to a receiving end. By means of carrying out compression processing on the collected speech information by using a low-bit-rate speech encoding algorithm, a code rate of speech information is greatly reduced, the capacity of the sent speech file is decreased, a bandwidth resource of a satellite mobile communication network is saved on, thus realizing a low-time delay and real-time talkback, solving the technical problem that a mobile terminal based on satellite mobile communications cannot use an instant communication application to realize a real-time speech talkback in the prior art, and improving the user experience.

Description

语音对讲方法、 装置和移动终端  Voice intercom method, device and mobile terminal
技术领域  Technical field
[0001] 本发明涉及通信技术领域, 特别是涉及到一种语音对讲方法、 装置和移动终端 背景技术  [0001] The present invention relates to the field of communications technologies, and in particular, to a voice intercom method, apparatus, and mobile terminal.
[0002] 利用地球静止轨道卫星或中、 低轨道卫星作为中继站, 实现区域乃至全球范围 的移动通信称为卫星移动通信。 它一般包括三部分: 通信卫星, 由一颗或多颗 卫星组成; 地面站, 包括系统控制中心和若干个信关站 (即把公共电话交换网 和移动用户连接起来的中转站) ; 移动用户通信终端, 包括车载、 舰载、 机载 终端和手持机。 用户可以在卫星波束的覆盖范围内自由移动, 卫星传递信号, 保持与地面通信系统和专用系统用户或其他移动用户的通信。  [0002] The use of geostationary orbit satellites or medium and low-orbit satellites as relay stations to achieve regional or global mobile communications is known as satellite mobile communications. It generally consists of three parts: a communication satellite, consisting of one or more satellites; a ground station, including a system control center and several gateway stations (ie, transit stations that connect the public switched telephone network to mobile users); mobile users Communication terminals, including onboard, shipboard, airborne terminals and handsets. The user can move freely within the coverage of the satellite beam, and the satellite transmits signals to maintain communication with terrestrial communication systems and dedicated system users or other mobile users.
[0003] 与其他通信方式相比, 卫星移动通信具有覆盖区域大、 通信距离远、 通信机动 灵活、 线路稳定可靠等优点。 因此, 卫星移动通信已经成为通信业务的一个重 要发展方向。  [0003] Compared with other communication methods, satellite mobile communication has the advantages of large coverage area, long communication distance, flexible communication and maneuverability, and stable and reliable lines. Therefore, satellite mobile communication has become an important development direction of communication services.
[0004] 随着卫星移动通信技术的迅速发展, 越来越多的移动终端支持卫星移动通信。  [0004] With the rapid development of satellite mobile communication technologies, more and more mobile terminals support satellite mobile communication.
与公众陆地移动通信网络一样, 卫星移动通信网络也支持 TCP/IP链路接入互联 网的功能, 因此移动终端可以通过卫星移动通信网络进行联网通信。 然而, 由 于卫星移动通信网络的带宽较窄, 因此无法利用即吋通信应用实现实吋的语音 对讲, 从而影响用户体验。  Like the public land mobile communication network, the satellite mobile communication network also supports the function of the TCP/IP link to access the Internet, so that the mobile terminal can communicate via the satellite mobile communication network. However, since the bandwidth of the satellite mobile communication network is narrow, it is impossible to implement an actual voice intercom using the instant communication application, thereby affecting the user experience.
技术问题  technical problem
[0005] 本发明的主要目的为提供一种语音对讲方法、 装置和移动终端, 旨在解决基于 卫星移动通信的移动终端无法利用即吋通信应用实现实吋的语音对讲的技术问 题。  [0005] A primary object of the present invention is to provide a voice intercom method, apparatus, and mobile terminal, which are directed to solving a technical problem that a mobile terminal based on satellite mobile communication cannot implement a real voice intercom using an instant communication application.
问题的解决方案  Problem solution
技术解决方案  Technical solution
[0006] 为达以上目的, 本发明实施例提出一种语音对讲方法, 所述方法包括以下步骤 [0007] 通过卫星移动通信网络与服务器保持长连接; [0006] In order to achieve the above objective, an embodiment of the present invention provides a voice intercom method, where the method includes the following steps. [0007] maintaining a long connection with the server through the satellite mobile communication network;
[0008] 利用低速率语音编码算法对采集的语音信息进行压缩处理并生成第一语音文件  [0008] compressing the collected voice information by using a low-rate speech coding algorithm and generating a first voice file
[0009] 向所述服务器发送所述第一语音文件, 以使所述服务器将所述第一语音文件发 送给对端。 Sending the first voice file to the server, so that the server sends the first voice file to the peer end.
[0010] 可选地, 所述通过卫星移动通信网络与服务器建立连接并保持长连接的步骤之 后还包括:  [0010] Optionally, after the step of establishing a connection with the server and maintaining a long connection through the satellite mobile communication network, the method further includes:
[0011] 从所述服务器获取对端发送的第二语音文件;  [0011] acquiring, by the server, a second voice file sent by the peer end;
[0012] 输出所述第二语音文件。 [0012] outputting the second voice file.
[0013] 可选地, 所述从所述服务器获取对端发送的第二语音文件的步骤包括:  [0013] Optionally, the step of obtaining, by the server, the second voice file sent by the peer end includes:
[0014] 接收所述服务器发送的所述第二语音文件的下载地址; [0014] receiving a download address of the second voice file sent by the server;
[0015] 根据所述下载地址下载所述第二语音文件。 [0015] downloading the second voice file according to the download address.
[0016] 可选地, 所述输出所述第二语音文件的步骤包括: [0016] Optionally, the step of outputting the second voice file includes:
[0017] 判断所述第二语音文件是否为低速率语音编码文件; [0017] determining whether the second voice file is a low rate voice encoded file;
[0018] 当为低速率语音编码文件吋, 利用低速率语音解码算法对所述第二语音文件进 行解码后予以播放。  [0018] When the file is a low rate speech code, the second speech file is decoded using a low rate speech decoding algorithm and played.
[0019] 可选地, 所述低速率语音编码算法为自适应多速率 AMR算法、 混合激励线性 预测编码 MELP算法、 码激励线性预测编码 CELP算法、 正弦变换编码 STC算法、 吋频域插值编码 TFI算法、 基音同步激励线性预测编码 PSELP算法、 多带激励编 码 MBE算法或波形内插编码 WI算法。  [0019] Optionally, the low-rate speech coding algorithm is an adaptive multi-rate AMR algorithm, a hybrid excitation linear predictive coding MELP algorithm, a code excitation linear predictive coding CELP algorithm, a sine transform coding STC algorithm, and a frequency domain interpolation coding TFI. Algorithm, pitch synchronization excitation linear prediction coding PSELP algorithm, multi-band excitation coding MBE algorithm or waveform interpolation coding WI algorithm.
[0020] 可选地, 当所述低速率语音压缩编码算法为 AMR算法吋, 所述利用低速率语 音压缩编码算法对采集的语音信息进行压缩处理并生成语音文件的步骤包括: [0020] Optionally, when the low-rate speech compression coding algorithm is an AMR algorithm, the step of compressing the collected voice information by using a low-rate speech compression coding algorithm and generating a voice file includes:
[0021] 利用所述 AMR算法对采集的语音信息进行压缩编码, 以降低所述语音信息的 码率至预设值, 并生成 AMR格式的语音文件。 [0021] The collected voice information is compression-encoded by using the AMR algorithm to reduce the bit rate of the voice information to a preset value, and generate a voice file in an AMR format.
[0022] 可选地, 所述预设值为 6.6kb/s。  [0022] Optionally, the preset value is 6.6 kb/s.
[0023] 可选地, 当所述低速率语音压缩编码算法为 MELP算法吋, 所述利用低速率语 音压缩编码算法对采集的语音信息进行压缩处理并生成语音文件的步骤包括: [0024] 利用所述 MELP算法对采集的语音信息进行压缩编码, 以降低所述语音信息的 码率至 2.4kb/s, 并生成 MELP格式的语音文件。 [0023] Optionally, when the low-rate voice compression coding algorithm is an MELP algorithm, the step of compressing the collected voice information by using a low-rate voice compression coding algorithm and generating a voice file includes: [0024] The collected voice information is compression-encoded by using the MELP algorithm to reduce the code rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format.
[0025] 可选地, 所述向所述服务器发送所述语音文件的步骤包括: [0025] Optionally, the step of sending the voice file to the server includes:
[0026] 采用 TCP/IP协议分包传输所述语音文件至所述服务器。 [0026] The voice file is transmitted to the server in a packet using a TCP/IP protocol.
[0027] 本发明实施例同吋提出一种语音对讲装置, 所述装置包括: [0027] The embodiment of the present invention provides a voice intercom device, and the device includes:
[0028] 连接模块, 用于通过卫星移动通信网络与服务器保持长连接; [0028] a connection module, configured to maintain a long connection with the server through the satellite mobile communication network;
[0029] 处理模块, 用于利用低速率语音编码算法对采集的语音信息进行压缩处理并生 成第一语音文件; [0029] a processing module, configured to compress the collected voice information by using a low-rate speech coding algorithm and generate a first voice file;
[0030] 发送模块, 用于向所述服务器发送所述第一语音文件, 以使所述服务器将所述 第一语音文件发送给对端。  And a sending module, configured to send the first voice file to the server, so that the server sends the first voice file to the peer end.
[0031] 可选地, 所述装置还包括: [0031] Optionally, the device further includes:
[0032] 获取模块, 用于从所述服务器获取对端发送的第二语音文件;  [0032] an obtaining module, configured to acquire, from the server, a second voice file sent by the peer end;
[0033] 输出模块, 用于输出所述第二语音文件。 [0033] an output module, configured to output the second voice file.
[0034] 可选地, 所述获取模块包括: [0034] Optionally, the acquiring module includes:
[0035] 接收单元, 用于接收所述服务器发送的所述第二语音文件的下载地址;  [0035] a receiving unit, configured to receive a download address of the second voice file sent by the server;
[0036] 下载单元, 用于根据所述下载地址下载所述第二语音文件。 [0036] a downloading unit, configured to download the second voice file according to the download address.
[0037] 可选地, 所述输出模块包括: [0037] Optionally, the output module includes:
[0038] 判断单元, 用于判断所述第二语音文件是否为低速率语音编码文件;  [0038] a determining unit, configured to determine whether the second voice file is a low-rate voice encoded file;
[0039] 播放单元, 用于当为低速率语音编码文件吋, 利用低速率语音解码算法对所述 第二语音文件进行解码后予以播放。  And a playing unit, configured to: when the file is a low-rate speech encoded file, play the second voice file by using a low-rate speech decoding algorithm.
[0040] 可选地, 当所述低速率语音压缩编码算法为 AMR算法吋, 所述处理模块用于[0040] Optionally, when the low-rate speech compression coding algorithm is an AMR algorithm, the processing module is used to
: 利用所述 AMR算法对采集的语音信息进行压缩编码, 以降低所述语音信息的 码率至预设值, 并生成 AMR格式的语音文件。 And compressing the collected voice information by using the AMR algorithm to reduce a bit rate of the voice information to a preset value, and generating a voice file in an AMR format.
[0041] 可选地, 当所述低速率语音压缩编码算法为 MELP算法吋, 所述处理模块用于[0041] Optionally, when the low-rate voice compression coding algorithm is a MELP algorithm, the processing module is used to
: 利用所述 MELP算法对采集的语音信息进行压缩编码, 以降低所述语音信息的 码率至 2.4kb/s, 并生成 MELP格式的语音文件。 And compressing the collected voice information by using the MELP algorithm to reduce the bit rate of the voice information to 2.4 kb/s, and generating a voice file in the MELP format.
[0042] 可选地, 所述发送模块用于: 采用 TCP/IP协议分包传输所述语音文件至所述服 务器。 [0043] 本发明还提出一种移动终端, 所述移动终端包括存储器、 处理器和至少一个被 存储在所述存储器中并被配置为由所述处理器执行的应用程序, 其特征在于, 所述应用程序被配置为用于执行前述语音对讲方法。 [0042] Optionally, the sending module is configured to: packet transmit the voice file to the server by using a TCP/IP protocol. The present invention also proposes a mobile terminal, the mobile terminal comprising a memory, a processor and at least one application stored in the memory and configured to be executed by the processor, characterized in that The application is configured to perform the aforementioned voice intercom method.
发明的有益效果  Advantageous effects of the invention
有益效果  Beneficial effect
[0044] 本发明实施例所提供的一种语音对讲方法, 通过卫星移动通信网络与服务器保 持长连接, 并利用低速率语音编码算法对采集的语音信息进行压缩处理, 从而 大大降低了语音信息的码率, 减小了发送的语音文件的容量, 节省了卫星移动 通信网络的带宽资源, 进而实现低吋延实吋对讲, 解决了现有技术中基于卫星 移动通信的移动终端无法利用即吋通信应用实现实吋的语音对讲的技术问题, 提升了用户体验。  [0044] A voice intercom method provided by an embodiment of the present invention maintains a long connection with a server through a satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing voice information. The code rate reduces the capacity of the transmitted voice file, saves the bandwidth resources of the satellite mobile communication network, and realizes the low-end, short-talk, intercom, and solves the problem that the mobile terminal based on satellite mobile communication in the prior art cannot be utilized. The communication application realizes the technical problem of the voice intercom and improves the user experience.
对附图的简要说明  Brief description of the drawing
附图说明  DRAWINGS
[0045] 图 1是本发明的语音对讲方法第一实施例的流程图;  1 is a flow chart of a first embodiment of a voice intercom method of the present invention;
[0046] 图 2是本发明的语音对讲方法第二实施例的流程图; 2 is a flow chart of a second embodiment of a voice intercom method of the present invention;
[0047] 图 3是本发明的语音对讲装置第一实施例的模块示意图; 3 is a block diagram showing a first embodiment of a voice intercom apparatus of the present invention;
[0048] 图 4是本发明的语音对讲装置第二实施例的模块示意图; 4 is a block diagram showing a second embodiment of a voice intercom apparatus of the present invention;
[0049] 图 5是图 4中语音对讲装置的获取模块的模块示意图; 5 is a block diagram of an acquisition module of the voice intercom device of FIG. 4;
[0050] 图 6是图 4中语音对讲装置的输出模块的模块示意图。 6 is a block diagram of an output module of the voice intercom device of FIG. 4.
[0051] 本发明目的的实现、 功能特点及优点将结合实施例, 参照附图做进一步说明。  [0051] The implementation, functional features, and advantages of the present invention will be further described with reference to the accompanying drawings.
实施该发明的最佳实施例  BEST MODE FOR CARRYING OUT THE INVENTION
本发明的最佳实施方式  BEST MODE FOR CARRYING OUT THE INVENTION
[0052] 应当理解, 此处所描述的具体实施例仅仅用以解释本发明, 并不用于限定本发 明。 The specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
[0053] 下面详细描述本发明的实施例, 所述实施例的示例在附图中示出, 其中自始至 终相同或类似的标号表示相同或类似的元件或具有相同或类似功能的元件。 下 面通过参考附图描述的实施例是示例性的, 仅用于解释本发明, 而不能解释为 对本发明的限制。 The embodiments of the present invention are described in detail below, and the examples of the embodiments are illustrated in the drawings, wherein the same or similar reference numerals are used to refer to the same or similar elements or elements having the same or similar functions. The embodiments described below with reference to the drawings are illustrative and are merely illustrative of the invention and are not construed as Limitations of the invention.
[0054] 本技术领域技术人员可以理解, 除非特意声明, 这里使用的单数形式"一"、 " 一个"、 "所述 "和"该"也可包括复数形式。 应该进一步理解的是, 本发明的说明 书中使用的措辞"包括"是指存在所述特征、 整数、 步骤、 操作、 元件和 /或组件 , 但是并不排除存在或添加一个或多个其他特征、 整数、 步骤、 操作、 元件、 组件和 /或它们的组。 应该理解, 当我们称元件被"连接"或"耦接"到另一元件吋 , 它可以直接连接或耦接到其他元件, 或者也可以存在中间元件。 此外, 这里 使用的"连接"或"耦接"可以包括无线连接或无线耦接。 这里使用的措辞 "和 /或"包 括一个或更多个相关联的列出项的全部或任一单元和全部组合。  The singular forms "a", "the", "the" and "the" It will be further understood that the phrase "comprising", used in the <RTI ID=0.0> </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> is intended to mean the presence of the features, integers, steps, operations, components and/or components, but does not exclude the presence or addition of one or more other features, Integers, steps, operations, components, components, and/or their groups. It will be understood that when we refer to an element being "connected" or "coupled" to another element, it can be directly connected or coupled to the other element, or an intermediate element can be present. Further, "connected" or "coupled" as used herein may include either a wireless connection or a wireless coupling. The phrase "and/or" used herein includes all or any of the elements and all combinations of one or more of the associated listed.
[0055] 本技术领域技术人员可以理解, 除非另外定义, 这里使用的所有术语 (包括技 术术语和科学术语) , 具有与本发明所属领域中的普通技术人员的一般理解相 同的意义。 还应该理解的是, 诸如通用字典中定义的那些术语, 应该被理解为 具有与现有技术的上下文中的意义一致的意义, 并且除非像这里一样被特定定 义, 否则不会用理想化或过于正式的含义来解释。  [0055] Those skilled in the art will appreciate that all terms (including technical and scientific terms) used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this invention belongs, unless otherwise defined. It should also be understood that terms such as those defined in a general dictionary should be understood to have meaning consistent with the meaning in the context of the prior art, and will not be idealized or excessive unless specifically defined as here. The formal meaning is explained.
[0056] 本技术领域技术人员可以理解, 这里所使用的 "终端"、 "终端设备"既包括无线 信号接收器的设备, 其仅具备无发射能力的无线信号接收器的设备, 又包括接 收和发射硬件的设备, 其具有能够在双向通信链路上, 执行双向通信的接收和 发射硬件的设备。 这种设备可以包括: 蜂窝或其他通信设备, 其具有单线路显 示器或多线路显示器或没有多线路显示器的蜂窝或其他通信设备; PCS (Persona 1 Communications Service, 个人通信系统) , 其可以组合语音、 数据处理、 传真 和 /或数据通信能力; PDA (Personal Digital Assistant, 个人数字助理) , 其可以 包括射频接收器、 寻呼机、 互联网 /内联网访问、 网络浏览器、 记事本、 日历和 / 或 GPS (Global Positioning System, 全球定位系统) 接收器; 常规膝上型和 /或掌 上型计算机或其他设备, 其具有和 /或包括射频接收器的常规膝上型和 /或掌上型 计算机或其他设备。 这里所使用的 "终端"、 "终端设备"可以是便携式、 可运输、 安装在交通工具 (航空、 海运和 /或陆地) 中的, 或者适合于和 /或配置为在本地 运行, 和 /或以分布形式, 运行在地球和 /或空间的任何其他位置运行。 这里所使 用的"终端"、 "终端设备"还可以是通信终端、 上网终端、 音乐 /视频播放终端, 例如可以是 PDA、 MID (Mobile Internet Device, 移动互联网设备) 和 /或具有音 乐 /视频播放功能的移动电话, 也可以是智能电视、 机顶盒等设备。 [0056] Those skilled in the art can understand that the "terminal" and "terminal device" used herein include both a device of a wireless signal receiver, a device having only a wireless signal receiver without a transmitting capability, and a receiving and receiving device. A device that transmits hardware having a receiving and transmitting hardware capable of performing two-way communication over a two-way communication link. Such a device may comprise: a cellular or other communication device having a single line display or a multi-line display or a cellular or other communication device without a multi-line display; PCS (Persona 1 Communications Service), which may combine voice, Data processing, fax and/or data communication capabilities; PDA (Personal Digital Assistant), which can include radio frequency receivers, pagers, Internet/Intranet access, web browsers, notepads, calendars and/or GPS ( Global Positioning System, Receiver; Conventional laptop and/or palmtop computer or other device having a conventional laptop and/or palmtop computer or other device that includes and/or includes a radio frequency receiver. As used herein, "terminal", "terminal device" may be portable, transportable, installed in a vehicle (aviation, sea and/or land), or adapted and/or configured to operate locally, and/or Run in any other location on the Earth and/or space in a distributed fashion. The "terminal" and "terminal device" used herein may also be a communication terminal, an internet terminal, a music/video playback terminal, For example, it may be a PDA, a MID (Mobile Internet Device), and/or a mobile phone with music/video playback function, or a smart TV, a set top box, or the like.
[0057] 本技术领域技术人员可以理解, 这里所使用的服务器, 其包括但不限于计算机 、 网络主机、 单个网络服务器、 多个网络服务器集或多个服务器构成的云。 在 此, 云由基于云计算 (Cloud Computing) 的大量计算机或网络服务器构成, 其 中, 云计算是分布式计算的一种, 由一群松散耦合的计算机集组成的一个超级 虚拟计算机。 本发明的实施例中, 服务器、 终端设备与 WNS服务器之间可通过 任何通信方式实现通信, 包括但不限于, 基于 3GPP、 LTE、 WIMAX的移动通信 、 基于 TCP/IP、 UDP协议的计算机网络通信以及基于蓝牙、 红外传输标准的近 距无线传输方式。 [0057] Those skilled in the art can understand that the server used herein includes, but is not limited to, a computer, a network host, a single network server, a plurality of network server sets, or a cloud composed of a plurality of servers. Here, the cloud consists of a large number of computers or network servers based on Cloud Computing, which is a kind of distributed computing, a super virtual computer composed of a group of loosely coupled computers. In the embodiment of the present invention, communication may be implemented by any communication means between the server, the terminal device and the WNS server, including but not limited to, mobile communication based on 3GPP, LTE, WIMAX, and computer network communication based on TCP/IP and UDP protocols. And short-range wireless transmission based on Bluetooth and infrared transmission standards.
[0058] 参照图 1, 提出本发明的语音对讲方法第一实施例, 所述方法包括以下步骤: [0059] Sl l、 通过卫星移动通信网络与服务器保持长连接。  Referring to FIG. 1, a first embodiment of a voice intercom method of the present invention is proposed. The method includes the following steps: [0059] Sl l, maintaining a long connection with a server through a satellite mobile communication network.
[0060] 本步骤 S11中, 移动终端通过卫星移动通信网络与服务器建立连接后, 则以一 定的心跳周期与服务器保持长连接, 即移动终端每隔一个心跳周期向服务器发 送一次心跳包, 以此保持二者之间的连接通路, 从而实现后续语音包的低吋延 实吋传送。  [0060] In this step S11, after the mobile terminal establishes a connection with the server through the satellite mobile communication network, the mobile terminal maintains a long connection with the server with a certain heartbeat period, that is, the mobile terminal sends a heartbeat packet to the server every other heartbeat period, thereby The connection path between the two is maintained, so that the subsequent voice packets are delayed and transmitted.
[0061] 可选地, 移动终端可以以预设的心跳周期与服务器保持长连接。  [0061] Optionally, the mobile terminal may maintain a long connection with the server in a preset heartbeat period.
[0062] 可选地, 移动终端可以根据参考心跳周期和卫星移动通信网络的信号质量对心 跳周期进行自适应调整, 获取周期较大且能保持稳定连接的自适应心跳周期, 并以自适应心跳周期与服务器保持长连接。 该参考心跳周期可以是预设的心跳 周期、 上一次连接吋使用的心跳周期、 其它移动通信网络 (如公众陆地移动通 信网络) 使用的心跳周期等。 [0062] Optionally, the mobile terminal may adaptively adjust the heartbeat period according to the reference heartbeat period and the signal quality of the satellite mobile communication network, acquire an adaptive heartbeat period with a large period and maintain a stable connection, and adopt an adaptive heartbeat The cycle maintains a long connection with the server. The reference heartbeat period may be a preset heartbeat period, a heartbeat period used by the last connection, a heartbeat period used by other mobile communication networks (such as the public land mobile communication network), and the like.
[0063] 举例而言, 移动终端首先以参考心跳周期进行长连接测试。 当参考心跳周期能 够维持长连接吋, 在参考心跳周期的基础上增加吋长进行长连接测试, 获取能 够维持长连接的自适应心跳周期, 例如: 在参考心跳周期的基础上逐次增加吋 长进行长连接测试, 直到不能维持长连接为止; 当不能维持长连接吋, 选取前 一次测试的心跳周期作为自适应心跳周期。 当参考心跳周期不能维持长连接吋 , 在参考心跳周期的基础上减少吋长进行长连接测试, 获取能够维持长连接的 自适应心跳周期, 例如: 在参考心跳周期的基础上逐次减少吋长进行长连接测 试, 直到能够维持长连接为止; 当能够维持长连接吋, 选取本次测试的心跳周 期作为自适应心跳周期。 最后, 移动终端以自适应心跳周期与服务器维持长连 接。 [0063] For example, the mobile terminal first performs a long connection test with a reference heartbeat period. When the reference heartbeat period can maintain a long connection, the length of the reference heartbeat period is increased to perform a long connection test to obtain an adaptive heartbeat period capable of maintaining a long connection, for example: increasing the length of the reference based on the reference heartbeat period Long connection test until the long connection cannot be maintained; When the long connection is not maintained, the heartbeat period of the previous test is selected as the adaptive heartbeat period. When the reference heartbeat period cannot maintain a long connection, the long connection test is performed on the basis of the reference heartbeat period to obtain a long connection test. The adaptive heartbeat period, for example: based on the reference heartbeat period, successively reduces the length of the long connection test until the long connection can be maintained; when the long connection is maintained, the heartbeat period of the test is selected as the adaptive heartbeat period. Finally, the mobile terminal maintains a long connection with the server in an adaptive heartbeat cycle.
[0064] 本发明实施例中, 移动终端可以是只支持卫星移动通信的卫星移动通信终端, 也可以既支持卫星移动通信又支持公众陆地移动通信的卫星移动通信和公众陆 地移动通信的融合终端。  In the embodiment of the present invention, the mobile terminal may be a satellite mobile communication terminal supporting only satellite mobile communication, or a fusion terminal supporting both satellite mobile communication and public land mobile communication for satellite mobile communication and public land mobile communication.
[0065] S12、 利用低速率语音编码算法对采集的语音信息进行处理并生成第一语音文 件。  [0065] S12. Process the collected voice information by using a low-rate speech coding algorithm and generate a first voice file.
[0066] 本发明实施例中, 移动终端可以通过微信、 易信、 QQ等即吋通讯应用与其它 终端进行语音对讲, 此吋其它终端则为该移动终端的对端。 当移动终端向对端 发送语音文件吋, 则该移动终端为发送端, 对端为接收端; 当移动终端接收对 端发送的语音文件吋, 则该移动终端为接收端, 对端为发送端。  In the embodiment of the present invention, the mobile terminal can perform voice intercom with other terminals through a short communication service such as WeChat, E-mail, and QQ, and the other terminals are the opposite ends of the mobile terminal. When the mobile terminal sends a voice file to the opposite end, the mobile terminal is the transmitting end, and the opposite end is the receiving end; when the mobile terminal receives the voice file sent by the opposite end, the mobile terminal is the receiving end, and the opposite end is the transmitting end. .
[0067] 当移动终端作为发送端吋, 则通过麦克风采集语音信息, 并利用低速率语音编 码算法对采集的语音信息进行处理, 生成第一语音文件。  [0067] When the mobile terminal is used as the transmitting end, the voice information is collected through the microphone, and the collected voice information is processed by using a low-rate voice encoding algorithm to generate a first voice file.
[0068] 可选地, 移动终端采集语音信息吋, 通过应用处理器采用 8位 ADC和 8k的采样 频率采集语音信息, 同吋对采集到的语音信息进行数字化录音。 采集的语音信 息的码率为 64kb/s。  [0068] Optionally, the mobile terminal collects voice information, and the application processor collects voice information by using an 8-bit ADC and a sampling frequency of 8 k, and digitally records the collected voice information. The code rate of the collected voice information is 64 kb/s.
[0069] 低速率语音编码算法可以是自适应多速率 (AMR) 算法、 混合激励线性预测 编码 (MELP) 算法、 码激励线性预测编码 (CELP) 算法、 正弦变换编码 (STC ) 算法、 吋频域插值编码 (TFI) 算法、 基音同步激励线性预测编码 (PSELP) 算法、 多带激励编码 (MBE) 算法、 波形内插编码 (WI) 算法等语音编码算法 中的任意一种。  [0069] The low rate speech coding algorithm may be an adaptive multi-rate (AMR) algorithm, a mixed excitation linear predictive coding (MELP) algorithm, a code excited linear predictive coding (CELP) algorithm, a sinusoidal transform coding (STC) algorithm, a chirp frequency domain. Any of the speech coding algorithms such as the Interpolation Coding (TFI) algorithm, the Pitch Synchronous Linear Predictive Coding (PSELP) algorithm, the Multi-Band Excitation Coding (MBE) algorithm, and the Waveform Interpolation Coding (WI) algorithm.
[0070] 例如, 以 AMR算法为例, 移动终端利用 AMR算法对采集的语音信息进行压缩 编码, 以降低语音信息的码率至预设值, 并生成 AMR格式的语音文件。 AMR可 采用从 6.6kb/s到 23.85kb/s共九种编码, 预设值优选最低码率 6.6kb/s。 从而大大降 低了语音信息的码率, 减小了语音文件的容量, 节省了卫星移动通信网络的带 宽资源, 进而实现低吋延实吋对讲。 [0071] 又如, 以 MELP算法为例, 移动终端利用 MELP算法对采集的语音信息进行压 缩编码, 以降低语音信息的码率至 2.4kb/s, 并生成 MELP格式的语音文件。 从而 大大降低了语音信息的码率, 减小了语音文件的容量, 节省了卫星移动通信网 络的带宽资源, 进而实现低吋延实吋对讲。 For example, taking the AMR algorithm as an example, the mobile terminal uses the AMR algorithm to compress and encode the collected voice information, so as to reduce the code rate of the voice information to a preset value, and generate a voice file in the AMR format. The AMR can use nine codes from 6.6 kb/s to 23.85 kb/s, and the preset value is preferably a minimum code rate of 6.6 kb/s. Thereby, the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized. For example, taking the MELP algorithm as an example, the mobile terminal uses the MELP algorithm to compress and encode the collected voice information to reduce the bit rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format. Thereby, the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized.
[0072] S13、 向服务器发送第一语音文件, 以使服务器将第一语音文件发送给对端。 [0072] S13. Send a first voice file to the server, so that the server sends the first voice file to the peer end.
[0073] 本发明实施例中, 移动终端的卫星通信调制解调器 (modem) 通过套接字 (so cket) 与服务器建立通信, 移动终端优选采用传输控制协议 /因特网互联协议 (T CP/IP) 协议分包传输语音文件至服务器。 也就是说, 移动终端将第一语音文件 分割成多个语音包, 按照顺序将多个语音包依次发送给服务器。 服务器接收到 多个语音包后, 按照语音包的起始标识和结束标识, 将语音包顺序存入缓存, 组成一个语音文件, 即还原出第一语音文件。 [0073] In the embodiment of the present invention, the satellite communication modem of the mobile terminal establishes communication with the server through a socket, and the mobile terminal preferably adopts a Transmission Control Protocol/Internet Protocol (T CP/IP) protocol. The package transmits voice files to the server. That is, the mobile terminal divides the first voice file into a plurality of voice packets, and sequentially transmits the plurality of voice packets to the server in sequence. After receiving the voice packets, the server stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the first voice file.
[0074] 每一个语音包均为一个 TCP/IP协议包, TCP/IP协议包的组成如下:  [0074] Each voice packet is a TCP/IP protocol packet, and the components of the TCP/IP protocol packet are as follows:
[0075] I…一包头一… μ…包体长度一-1 包体 1 [0075] I...one head one... μ...body length one-1 body 1
[0076] 移动终端与服务器可以对包头的定义进行约定 (如设置不同的标识) , 服务器 对 TCP/IP协议包的包头进行解析, 区分出传送语音文件的网络为卫星移动通信 网络还是公众陆地移动通信网络, 也就是说, 区分出发送端发送的语音文件为 低速率语音编码文件还是普通语音编码文件。  [0076] The mobile terminal and the server may agree on the definition of the packet header (such as setting different identifiers), and the server parses the packet header of the TCP/IP protocol packet to distinguish whether the network transmitting the voice file is a satellite mobile communication network or a public land mobile. The communication network, that is, distinguishes whether the voice file sent by the sender is a low rate voice code file or a normal voice code file.
[0077] 服务器可以采取支持多个客户端并发接入的软件架构, 比如 MINA、 Erlang等 [0077] The server may adopt a software architecture that supports concurrent access of multiple clients, such as MINA, Erlang, etc.
, 支持多用户高并发访问服务器。 例如, 采用多线程机制, 一个线程用于监听 客户请求, 多个线程用于处理多个用户并发请求。 , Support multi-user high concurrent access to the server. For example, with a multi-threading mechanism, one thread is used to listen for client requests, and multiple threads are used to handle multiple concurrent requests.
[0078] 服务器端的具体流程为: 创建服务器端对象产生监听线程, 启动端口侦听, 幵 启接受客户端连接请求, 当有客户端连接上来吋, 创建客户端对象产生新线程 。 向客户端发送数据, 创建数据流传输对象, 启动数据侦听, 当接收到数据吋 , 判断数据长度。 当数据长度为 0吋, 判定连接已断幵, 刪除客户端对象和无用 线程; 当数据长度不为 0吋, 处理该数据。  [0078] The specific process on the server side is: creating a server-side object to generate a listening thread, starting port listening, and accepting a client connection request, when a client connection is up, creating a client object to generate a new thread. Send data to the client, create a data stream transfer object, start data snooping, and when receiving data, determine the data length. When the data length is 0吋, it is determined that the connection has been broken, and the client object and the useless thread are deleted; when the data length is not 0吋, the data is processed.
[0079] 服务器接收到第一语音文件后, 可以采用以下两种方式向接收端发送该语音文 件: 一种是向接收端发送第一语音文件的下载地址, 以使接收端根据下载地址 直接下载第一语音文件; 另一种是采用 TCP/IP协议分包传输第一语音文件至接 收端。 [0079] After receiving the first voice file, the server may send the voice file to the receiving end in the following two ways: One is to send the download address of the first voice file to the receiving end, so that the receiving end directly downloads according to the download address. The first voice file; the other is to use the TCP/IP protocol to sub-packet the first voice file to the next Received.
[0080] 如果接收端接入的是卫星移动通信网络, 优先采用下载的方式, 可以减小吋延 。 如果接收端接入的是公众陆地移动通信网络, 则两种方式均可。  [0080] If the receiving end accesses the satellite mobile communication network, the downloading method is preferred, and the delay can be reduced. If the receiving end accesses the public land mobile communication network, both methods are available.
[0081] 本发明实施例的语音对讲方法, 通过卫星移动通信网络与服务器保持长连接, 并利用低速率语音编码算法对采集的语音信息进行压缩处理, 从而大大降低了 语音信息的码率, 减小了发送的语音文件的容量, 节省了卫星移动通信网络的 带宽资源, 进而实现低吋延实吋对讲, 解决了现有技术中基于卫星移动通信的 移动终端无法利用即吋通信应用实现实吋的语音对讲的技术问题, 提升了用户 体验。  The voice intercom method of the embodiment of the present invention maintains a long connection with the server through the satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing the bit rate of the voice information. The capacity of the transmitted voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized, and the mobile terminal based on the satellite mobile communication in the prior art cannot be solved by using the instant communication application. The technical problems of the actual voice intercom have improved the user experience.
[0082] 进一步地, 如图 2所示, 在本发明的语音对讲方法第二实施例中, 当移动终端 作为接收端吋, 步骤 S11之后还包括:  [0082] Further, as shown in FIG. 2, in the second embodiment of the voice intercom method of the present invention, when the mobile terminal is the receiving end, step S11 further includes:
[0083] S14、 从服务器获取对端发送的第二语音文件。 [0083] S14. Acquire a second voice file sent by the peer end from the server.
[0084] 本发明实施例中, 服务器优选向移动终端发送第二语音文件的下载地址, 移动 终端接收服务器发送的下载地址, 并根据下载地址下载第二语音文件。 通过下 载方式获取第二语音文件, 可以减小语音对讲的吋延, 提升用户体验。  In the embodiment of the present invention, the server preferably sends the download address of the second voice file to the mobile terminal, the mobile terminal receives the download address sent by the server, and downloads the second voice file according to the download address. By obtaining the second voice file in the download mode, the delay of the voice intercom can be reduced, and the user experience is improved.
[0085] 在其它实施例中, 服务器也可以采用 TCP/IP协议向移动终端分包传输第二语音 文件, 即服务器将第二语音文件分割成多个语音包, 按照顺序将多个语音包依 次发送给移动终端。 移动终端接收到多个语音包后, 按照语音包的起始标识和 结束标识, 将语音包顺序存入缓存, 组成一个语音文件, 即还原出第二语音文 件。  [0085] In other embodiments, the server may also use the TCP/IP protocol to packetize and transmit the second voice file to the mobile terminal, that is, the server divides the second voice file into multiple voice packets, and sequentially sequentially multiple voice packets. Send to the mobile terminal. After receiving the voice packets, the mobile terminal stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the second voice file.
[0086] S15、 输出第二语音文件。  [0086] S15. Output a second voice file.
[0087] 本发明实施例中, 移动终端接收到第二语音文件后, 首先判断第二语音文件是 否为低速率语音编码文件; 当为低速率语音编码文件吋, 则通过低速率语音解 码器利用低速率语音解码算法对第二语音文件进行解码后予以播放; 当为普通 语音编码文件吋, 则通过宽带语音解码器对第二语音文件进行解码后予以播放  In the embodiment of the present invention, after receiving the second voice file, the mobile terminal first determines whether the second voice file is a low-rate voice code file; when the voice code file is a low-rate voice code file, the mobile terminal uses the low-rate voice decoder. The low-rate speech decoding algorithm decodes the second speech file and plays it; when it is a normal speech-encoded file, the second speech file is decoded by the wide-band speech decoder and then played.
[0088] 移动终端可以通过第二语音文件的标识信息来判断其是否为低速率语音编码文 件, 该标识信息可以设置于第二语音文件的语音包的包头。 [0089] 例如, 当第二语音文件的标识信息为第一标识吋, 判定该第二语音文件为低速 率语音编码文件; 当第二语音文件的标识信息为第二标识吋, 判定该第二语音 文件为普通语音编码文件。 [0088] The mobile terminal may determine whether it is a low-rate voice encoded file by using the identifier information of the second voice file, and the identifier information may be set in a packet header of the voice packet of the second voice file. [0089] For example, when the identification information of the second voice file is the first identifier, the second voice file is determined to be a low-rate voice encoded file; and when the identifier information of the second voice file is the second identifier, determining the second The voice file is a normal voice encoded file.
[0090] 又如, 当第二语音文件的标识信息为第一标识吋, 判定该第二语音文件为低速 率语音编码文件; 当第二语音文件的标识信息为空 (即没有标识) 吋, 判定该 第二语音文件为普通语音编码文件。 或者, 反之亦可。 [0090] For example, when the identification information of the second voice file is the first identifier, determining that the second voice file is a low-rate voice encoded file; when the identifier information of the second voice file is empty (ie, no identifier), It is determined that the second voice file is a normal voice encoded file. Or, vice versa.
[0091] 本实施例的语音对讲方法, 通过下载方式获取第二语音文件, 减小了语音对讲 的吋延, 提升了用户体验。 通过利用低速率语音解码算法对第二语音文件进行 解码, 实现了与接入卫星移动通信网络的移动终端进行语音对讲。 [0091] The voice intercom method of the embodiment obtains the second voice file by using the download mode, which reduces the delay of the voice intercom and improves the user experience. By using the low-rate speech decoding algorithm to decode the second speech file, a voice intercom is performed with the mobile terminal accessing the satellite mobile communication network.
[0092] 本发明实施例中, 接入卫星移动通信网络的移动终端, 既可以与接入卫星移动 通信网络的其它移动终端进行语音对讲, 又可以与接入公众陆地移动通信网络 的其它移动终端进行语音对讲。 [0092] In the embodiment of the present invention, a mobile terminal accessing a satellite mobile communication network can perform voice intercommunication with other mobile terminals accessing the satellite mobile communication network, and can also access other mobiles of the public land mobile communication network. The terminal performs voice intercom.
[0093] 参照图 3, 提出本发明的语音对讲装置第一实施例, 所述装置应用于移动终端Referring to FIG. 3, a first embodiment of a voice intercom apparatus of the present invention is proposed, and the apparatus is applied to a mobile terminal.
, 当然也可以应用于其它的终端设备, 所述装置包括连接模块 10、 处理模块 20 和发送模块 30, 其中: Of course, the device can also be applied to other terminal devices, and the device includes a connection module 10, a processing module 20, and a sending module 30, where:
[0094] 连接模块 10: 用于通过卫星移动通信网络与服务器保持长连接。 [0094] Connection module 10: for maintaining a long connection with the server via the satellite mobile communication network.
[0095] 本发明实施例中, 连接模块 10通过卫星移动通信网络与服务器建立连接后, 则 以一定的心跳周期与服务器保持长连接, 即连接模块 10每隔一个心跳周期向服 务器发送一次心跳包, 以此保持二者之间的连接通路, 从而实现后续语音包的 低吋延实吋传送。 In the embodiment of the present invention, after the connection module 10 establishes a connection with the server through the satellite mobile communication network, the connection module 10 maintains a long connection with the server with a certain heartbeat period, that is, the connection module 10 sends a heartbeat packet to the server every other heartbeat period. In order to maintain the connection path between the two, the low-speed transmission and subsequent transmission of the subsequent voice packets are realized.
[0096] 可选地, 连接模块 10可以以预设的心跳周期与服务器保持长连接。  [0096] Optionally, the connection module 10 may maintain a long connection with the server in a preset heartbeat period.
[0097] 可选地, 连接模块 10可以根据参考心跳周期和卫星移动通信网络的信号质量对 心跳周期进行自适应调整, 获取周期较大且能保持稳定连接的自适应心跳周期 , 并以自适应心跳周期与服务器保持长连接。 该参考心跳周期可以是预设的心 跳周期、 上一次连接吋使用的心跳周期、 其它移动通信网络 (如公众陆地移动 通信网络) 使用的心跳周期等。 [0097] Optionally, the connection module 10 can adaptively adjust the heartbeat period according to the reference heartbeat period and the signal quality of the satellite mobile communication network, and acquire an adaptive heartbeat period with a large period and maintaining a stable connection, and adaptively The heartbeat cycle stays connected to the server. The reference heartbeat period may be a preset heartbeat period, a heartbeat period used by the last connection, a heartbeat period used by other mobile communication networks (such as a public land mobile communication network), and the like.
[0098] 举例而言, 连接模块 10首先以参考心跳周期进行长连接测试。 当参考心跳周期 能够维持长连接吋, 在参考心跳周期的基础上增加吋长进行长连接测试, 获取 能够维持长连接的自适应心跳周期, 例如: 在参考心跳周期的基础上逐次增加 吋长进行长连接测试, 直到不能维持长连接为止; 当不能维持长连接吋, 选取 前一次测试的心跳周期作为自适应心跳周期。 当参考心跳周期不能维持长连接 吋, 在参考心跳周期的基础上减少吋长进行长连接测试, 获取能够维持长连接 的自适应心跳周期, 例如: 在参考心跳周期的基础上逐次减少吋长进行长连接 测试, 直到能够维持长连接为止; 当能够维持长连接吋, 选取本次测试的心跳 周期作为自适应心跳周期。 最后, 连接模块 10以自适应心跳周期与服务器维持 长连接。 [0098] For example, the connection module 10 first performs a long connection test with a reference heartbeat cycle. When the reference heartbeat period can maintain a long connection, the length of the reference heartbeat period is increased to perform a long connection test. An adaptive heartbeat period capable of maintaining a long connection, for example: increasing the length of the long connection test on the basis of the reference heartbeat period until the long connection cannot be maintained; when the long connection cannot be maintained, the heartbeat period of the previous test is selected as Adaptive heartbeat cycle. When the reference heartbeat period cannot maintain a long connection, the long connection test is performed on the basis of the reference heartbeat period to obtain an adaptive heartbeat period capable of maintaining a long connection, for example, the length of the reference is continuously reduced based on the reference heartbeat period. Long connection test until long connection can be maintained; When long connection is maintained, the heartbeat period of this test is selected as the adaptive heartbeat period. Finally, the connection module 10 maintains a long connection with the server in an adaptive heartbeat cycle.
[0099] 本发明实施例中, 移动终端可以是只支持卫星移动通信的卫星移动通信终端, 也可以既支持卫星移动通信又支持公众陆地移动通信的卫星移动通信和公众陆 地移动通信的融合终端。  In the embodiment of the present invention, the mobile terminal may be a satellite mobile communication terminal supporting only satellite mobile communication, or a fusion terminal supporting both satellite mobile communication and public land mobile communication for satellite mobile communication and public land mobile communication.
[0100] 处理模块 20: 用于利用低速率语音编码算法对采集的语音信息进行压缩处理并 生成第一语音文件。  [0100] The processing module 20 is configured to perform compression processing on the collected voice information by using a low-rate voice coding algorithm and generate a first voice file.
[0101] 本发明实施例中, 语音对讲装置可以通过微信、 易信、 QQ等即吋通讯应用与 其它终端进行语音对讲, 此吋其它终端则为该移动终端的对端。 当移动终端向 对端发送语音文件吋, 则该移动终端为发送端, 对端为接收端; 当移动终端接 收对端发送的语音文件吋, 则该移动终端为接收端, 对端为发送端。  In the embodiment of the present invention, the voice intercom device can perform voice intercom with other terminals through a short communication service such as WeChat, E-mail, and QQ, and the other terminals are the opposite ends of the mobile terminal. When the mobile terminal sends a voice file to the opposite end, the mobile terminal is the transmitting end, and the opposite end is the receiving end; when the mobile terminal receives the voice file sent by the opposite end, the mobile terminal is the receiving end, and the opposite end is the transmitting end. .
[0102] 当移动终端作为发送端吋, 处理模块 20则通过麦克风采集语音信息, 并利用低 速率语音编码算法对采集的语音信息进行处理, 生成第一语音文件。  [0102] When the mobile terminal is used as the transmitting end, the processing module 20 collects the voice information through the microphone, and processes the collected voice information by using a low-rate voice encoding algorithm to generate a first voice file.
[0103] 可选地, 处理模块 20采集语音信息吋, 通过应用处理器采用 8位 ADC和 8k的采 样频率采集语音信息, 同吋对采集到的语音信息进行数字化录音。 采集的语音 信息的码率为 64kb/s。  [0103] Optionally, the processing module 20 collects voice information, and uses the 8-bit ADC and the 8K sampling frequency to collect voice information, and digitally records the collected voice information. The code rate of the collected voice information is 64 kb/s.
[0104] 低速率语音编码算法可以是自适应多速率 (AMR) 算法、 混合激励线性预测 编码 (MELP) 算法、 码激励线性预测编码 (CELP) 算法、 正弦变换编码 (STC ) 算法、 吋频域插值编码 (TFI) 算法、 基音同步激励线性预测编码 (PSELP) 算法、 多带激励编码 (MBE) 算法、 波形内插编码 (WI) 算法等语音编码算法 中的任意一种。  [0104] The low-rate speech coding algorithm may be an adaptive multi-rate (AMR) algorithm, a mixed excitation linear predictive coding (MELP) algorithm, a code excited linear predictive coding (CELP) algorithm, a sinusoidal transform coding (STC) algorithm, and a frequency domain. Any of the speech coding algorithms such as the Interpolation Coding (TFI) algorithm, the Pitch Synchronous Linear Predictive Coding (PSELP) algorithm, the Multi-Band Excitation Coding (MBE) algorithm, and the Waveform Interpolation Coding (WI) algorithm.
[0105] 例如, 以 AMR算法为例, 处理模块 20利用 AMR算法对采集的语音信息进行压 缩编码, 以降低语音信息的码率至预设值, 并生成 AMR格式的语音文件。 AMR 可采用从 6.6kb/s到 23.85kb/s共九种编码, 预设值优选最低码率 6.6kb/s。 从而大大 降低了语音信息的码率, 减小了语音文件的容量, 节省了卫星移动通信网络的 带宽资源, 进而实现低吋延实吋对讲。 [0105] For example, taking the AMR algorithm as an example, the processing module 20 uses the AMR algorithm to compress the collected voice information. Reduce encoding to reduce the bit rate of voice information to a preset value, and generate a voice file in AMR format. The AMR can use nine codes from 6.6 kb/s to 23.85 kb/s, and the preset value is preferably a minimum code rate of 6.6 kb/s. Thereby, the code rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized.
[0106] 又如, 以 MELP算法为例, 处理模块 20利用 MELP算法对采集的语音信息进行 压缩编码, 以降低语音信息的码率至 2.4kb/s, 并生成 MELP格式的语音文件。 从 而大大降低了语音信息的码率, 减小了语音文件的容量, 节省了卫星移动通信 网络的带宽资源, 进而实现低吋延实吋对讲。  For example, taking the MELP algorithm as an example, the processing module 20 uses the MELP algorithm to compress and encode the collected voice information to reduce the bit rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format. Therefore, the bit rate of the voice information is greatly reduced, the capacity of the voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-end, short-talking intercom is realized.
[0107] 发送模块 30: 用于向服务器发送第一语音文件, 以使服务器将第一语音文件发 送给对端。  [0107] The sending module 30 is configured to send the first voice file to the server, so that the server sends the first voice file to the peer end.
[0108] 本发明实施例中, 发送模块 30优选采用 TCP/IP协议分包传输语音文件至服务器 [0108] In the embodiment of the present invention, the sending module 30 preferably uses a TCP/IP protocol to packetize and transmit voice files to the server.
。 也就是说, 发送模块 30将第一语音文件分割成多个语音包, 按照顺序将多个 语音包依次发送给服务器。 服务器接收到多个语音包后, 按照语音包的起始标 识和结束标识, 将语音包顺序存入缓存, 组成一个语音文件, 即还原出第一语 音文件。 . That is, the transmitting module 30 divides the first voice file into a plurality of voice packets, and sequentially transmits the plurality of voice packets to the server in sequence. After receiving the voice packets, the server stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, the first voice file is restored.
[0109] 每一个语音包均为一个 TCP/IP协议包, TCP/IP协议包的组成如下:  [0109] Each voice packet is a TCP/IP protocol packet, and the components of the TCP/IP protocol packet are as follows:
[0110] I…一包头―—— μ…包体长度—— I 包体 1  [0110] I... a baotou --- μ...body length - I package 1
[0111] 发送模块 30与服务器可以对包头的定义进行约定 (如设置不同的标识) , 服务 器对 TCP/IP协议包的包头进行解析, 区分出传送语音文件的网络为卫星移动通 信网络还是公众陆地移动通信网络, 也就是说, 区分出发送端发送的语音文件 为低速率语音编码文件还是普通语音编码文件。  [0111] The sending module 30 and the server may agree on the definition of the packet header (such as setting different identifiers), and the server parses the packet header of the TCP/IP protocol packet to distinguish whether the network transmitting the voice file is a satellite mobile communication network or a public land. The mobile communication network, that is, distinguishes whether the voice file sent by the sender is a low rate voice code file or a normal voice code file.
[0112] 本发明实施例的语音对讲装置, 通过卫星移动通信网络与服务器保持长连接, 并利用低速率语音编码算法对采集的语音信息进行压缩处理, 从而大大降低了 语音信息的码率, 减小了发送的语音文件的容量, 节省了卫星移动通信网络的 带宽资源, 进而实现低吋延实吋对讲, 解决了现有技术中基于卫星移动通信的 移动终端无法利用即吋通信应用实现实吋的语音对讲的技术问题, 提升了用户 体验。  [0112] The voice intercom device in the embodiment of the present invention maintains a long connection with the server through the satellite mobile communication network, and compresses the collected voice information by using a low-rate speech coding algorithm, thereby greatly reducing the bit rate of the voice information. The capacity of the transmitted voice file is reduced, the bandwidth resource of the satellite mobile communication network is saved, and the low-definition and short-talking intercom is realized, and the mobile terminal based on the satellite mobile communication in the prior art cannot be solved by using the instant communication application. The technical problems of the actual voice intercom have improved the user experience.
[0113] 进一步地, 如图 4所示, 在本发明的语音对讲装置第二实施例中, 该装置还包 括获取模块 40和输出模块 50, 获取模块 40用于从服务器获取对端发送的第二语 音文件, 输出模块 50用于输出第二语音文件。 [0113] Further, as shown in FIG. 4, in the second embodiment of the voice intercom apparatus of the present invention, the apparatus further includes The obtaining module 40 is configured to acquire the second voice file sent by the peer end, and the output module 50 is configured to output the second voice file.
[0114] 本发明实施例中, 服务器优选向移动终端发送第二语音文件的下载地址。 此吋 , 获取模块 40如图 5所示, 包括接收单元 41和下载单元 42, 接收单元 41用于接收 服务器发送的第二语音文件的下载地址, 下载单元 42用于根据下载地址下载第 二语音文件。 [0114] In the embodiment of the present invention, the server preferably sends the download address of the second voice file to the mobile terminal. As shown in FIG. 5, the obtaining module 40 includes a receiving unit 41 and a downloading unit 42. The receiving unit 41 is configured to receive a download address of the second voice file sent by the server, and the downloading unit 42 is configured to download the second voice according to the download address. file.
[0115] 在其它实施例中, 服务器也可以采用 TCP/IP协议向移动终端分包传输第二语音 文件, 即服务器将第二语音文件分割成多个语音包, 按照顺序将多个语音包依 次发送给移动终端。 获取模块 40接收多个语音包, 并按照语音包的起始标识和 结束标识, 将语音包顺序存入缓存, 组成一个语音文件, 即还原出第二语音文 件。  [0115] In other embodiments, the server may also use the TCP/IP protocol to packetize and transmit the second voice file to the mobile terminal, that is, the server divides the second voice file into multiple voice packets, and sequentially sequentially multiple voice packets. Send to the mobile terminal. The obtaining module 40 receives the plurality of voice packets, and stores the voice packets in the cache according to the start identifier and the end identifier of the voice packet to form a voice file, that is, restores the second voice file.
[0116] 如图 6所示, 输出模块 50包括判断单元 51和播放单元 52, 其中: 判断单元 51用 于判断第二语音文件是否为低速率语音编码文件; 播放单元 52用于当第二语音 文件为低速率语音编码文件吋, 通过低速率语音解码器利用低速率语音解码算 法对第二语音文件进行解码后予以播放; 当第二语音文件为普通语音编码文件 吋, 通过宽带语音解码器对第二语音文件进行解码后予以播放。  As shown in FIG. 6, the output module 50 includes a determining unit 51 and a playing unit 52, wherein: the determining unit 51 is configured to determine whether the second voice file is a low-rate voice encoded file; and the playing unit 52 is configured to use the second voice. The file is a low-rate speech encoded file, and the second speech file is decoded by a low-rate speech decoder using a low-rate speech decoding algorithm; when the second speech file is a normal speech-encoded file, the broadband speech decoder is used. The second voice file is decoded and played.
[0117] 判断单元 51可以通过第二语音文件的标识信息来判断其是否为低速率语音编码 文件, 该标识信息可以设置于第二语音文件的语音包的包头。  [0117] The determining unit 51 may determine whether it is a low-rate voice encoding file by using the identification information of the second voice file, and the identifier information may be set in the packet header of the voice packet of the second voice file.
[0118] 例如, 当第二语音文件的标识信息为第一标识吋, 判断单元 51判定该第二语音 文件为低速率语音编码文件; 当第二语音文件的标识信息为第二标识吋, 判断 单元 51判定该第二语音文件为普通语音编码文件。  [0118] For example, when the identification information of the second voice file is the first identifier, the determining unit 51 determines that the second voice file is a low-rate voice encoded file; and when the identifier information of the second voice file is the second identifier, determining Unit 51 determines that the second voice file is a normal voice encoded file.
[0119] 又如, 当第二语音文件的标识信息为第一标识吋, 判断单元 51判定该第二语音 文件为低速率语音编码文件; 当第二语音文件的标识信息为空 (即没有标识) 吋, 判断单元 51判定该第二语音文件为普通语音编码文件。 或者, 反之亦可。  [0119] For example, when the identification information of the second voice file is the first identifier, the determining unit 51 determines that the second voice file is a low-rate voice encoded file; when the identifier information of the second voice file is empty (ie, there is no identifier)吋, the determining unit 51 determines that the second voice file is a normal voice encoded file. Or, vice versa.
[0120] 本实施例的语音对讲装置, 通过下载方式获取第二语音文件, 减小了语音对讲 的吋延, 提升了用户体验。 通过利用低速率语音解码算法对第二语音文件进行 解码, 实现了与接入卫星移动通信网络的移动终端进行语音对讲。  [0120] The voice intercom device of the embodiment obtains the second voice file by using the download mode, which reduces the delay of the voice intercom and improves the user experience. By using the low-rate speech decoding algorithm to decode the second speech file, a voice intercom is performed with the mobile terminal accessing the satellite mobile communication network.
[0121] 本发明同吋提出一种移动终端, 所述移动终端包括存储器、 处理器和至少一个 被存储在所述存储器中并被配置为由所述处理器执行的应用程序, 所述应用程 序被配置为用于执行语音对讲方法。 所述语音对讲方法包括以下步骤: 通过卫 星移动通信网络与服务器保持长连接; 利用低速率语音编码算法对采集的语音 信息进行压缩处理并生成第一语音文件; 向所述服务器发送所述第一语音文件 , 以使所述服务器将所述第一语音文件发送给接收端。 本实施例中所描述的语 音对讲方法为本发明中上述实施例所涉及的语音对讲方法, 在此不再赘述。 [0121] The present invention also proposes a mobile terminal, where the mobile terminal includes a memory, a processor, and at least one An application stored in the memory and configured to be executed by the processor, the application being configured to execute a voice intercom method. The voice intercom method includes the following steps: maintaining a long connection with a server through a satellite mobile communication network; compressing the collected voice information by using a low rate speech coding algorithm to generate a first voice file; and transmitting the first message to the server a voice file, such that the server sends the first voice file to the receiving end. The voice intercom method described in this embodiment is the voice intercom method involved in the foregoing embodiment of the present invention, and details are not described herein again.
[0122] 本领域技术人员可以理解, 本发明包括涉及用于执行本申请中所述操作中的一 项或多项的设备。 这些设备可以为所需的目的而专门设计和制造, 或者也可以 包括通用计算机中的已知设备。 这些设备具有存储在其内的计算机程序, 这些 计算机程序选择性地激活或重构。 这样的计算机程序可以被存储在设备 (例如 , 计算机) 可读介质中或者存储在适于存储电子指令并分别耦联到总线的任何 类型的介质中, 所述计算机可读介质包括但不限于任何类型的盘 (包括软盘、 硬盘、 光盘、 CD-ROM、 和磁光盘) 、 ROM (Read-Only Memory , 只读存储器 ) 、 RAM (Random Access Memory , 随机存储器) 、 EPROM (Erasable Programmable Read-Only [0122] Those skilled in the art will appreciate that the present invention includes apparatus related to performing one or more of the operations described herein. These devices may be specially designed and manufactured for the required purposes, or may also include known devices in a general purpose computer. These devices have computer programs stored therein that are selectively activated or reconfigured. Such computer programs may be stored in a device (eg, computer) readable medium or in any type of medium suitable for storing electronic instructions and respectively coupled to a bus, including but not limited to any Types of disks (including floppy disks, hard disks, CDs, CD-ROMs, and magneto-optical disks), ROM (Read-Only Memory), RAM (Random Access Memory), EPROM (Erasable Programmable Read-Only)
Memory , 可擦写可编程只读存储器) 、 EEPROM (Electrically Erasable Programmable Read-Only Memory , 电可擦可编程只读存储器) 、 闪存、 磁性卡 片或光线卡片。 也就是, 可读介质包括由设备 (例如, 计算机) 以能够读的形 式存储或传输信息的任何介质。  Memory, EEPROM (Electrically Erasable Programmable Read-Only Memory), flash memory, magnetic card or light card. That is, a readable medium includes any medium that is stored or transmitted by a device (e.g., a computer) in a readable form.
[0123] 本技术领域技术人员可以理解, 可以用计算机程序指令来实现这些结构图和 / 或框图和 /或流图中的每个框以及这些结构图和 /或框图和 /或流图中的框的组合。 本技术领域技术人员可以理解, 可以将这些计算机程序指令提供给通用计算机 、 专业计算机或其他可编程数据处理方法的处理器来实现, 从而通过计算机或 其他可编程数据处理方法的处理器来执行本发明公幵的结构图和 /或框图和 /或流 图的框或多个框中指定的方案。  [0123] Those skilled in the art will appreciate that each block of the block diagrams and/or block diagrams and/or flow diagrams can be implemented by computer program instructions, as well as in the block diagrams and/or block diagrams and/or flow diagrams. The combination of boxes. Those skilled in the art will appreciate that these computer program instructions can be implemented by a general purpose computer, a professional computer, or a processor of other programmable data processing methods, such that the processor is executed by a computer or other programmable data processing method. The block diagrams and/or block diagrams of the invention and/or the schemes specified in the blocks or blocks of the flow diagram are invented.
[0124] 本技术领域技术人员可以理解, 本发明中已经讨论过的各种操作、 方法、 流程 中的步骤、 措施、 方案可以被交替、 更改、 组合或刪除。 进一步地, 具有本发 明中已经讨论过的各种操作、 方法、 流程中的其他步骤、 措施、 方案也可以被 交替、 更改、 重排、 分解、 组合或刪除。 进一步地, 现有技术中的具有与本发 明中公幵的各种操作、 方法、 流程中的步骤、 措施、 方案也可以被交替、 更改 、 重排、 分解、 组合或刪除。 [0124] Those skilled in the art can understand that the various operations, methods, and steps, measures, and solutions in the present invention may be alternated, changed, combined, or deleted. Further, various operations, methods, and other steps, measures, and solutions in the process that have been discussed in the present invention may also be Alternate, change, rearrange, decompose, combine, or delete. Further, the steps, measures, and solutions in the various operations, methods, and processes in the prior art may be alternated, changed, rearranged, decomposed, combined, or deleted.
以上所述仅为本发明的优选实施例, 并非因此限制本发明的专利范围, 凡是利 用本发明说明书及附图内容所作的等效结构或等效流程变换, 或直接或间接运 用在其他相关的技术领域, 均同理包括在本发明的专利保护范围内。  The above description is only the preferred embodiment of the present invention, and is not intended to limit the scope of the invention, and the equivalent structure or equivalent flow transformation made by the specification and the drawings of the present invention may be directly or indirectly applied to other related The technical field is equally included in the scope of patent protection of the present invention.

Claims

权利要求书 Claim
[权利要求 1] 一种语音对讲方法, 其特征在于, 包括以下步骤:  [Claim 1] A voice intercom method, comprising the steps of:
通过卫星移动通信网络与服务器保持长连接;  Maintaining a long connection to the server via a satellite mobile communication network;
利用低速率语音编码算法对采集的语音信息进行压缩处理并, 生成第 一语音文件;  Using the low-rate speech coding algorithm to compress the collected speech information and generate a first speech file;
向所述服务器发送所述第一语音文件, 以使所述服务器将所述第一语 音文件发送给对端。  Sending the first voice file to the server, so that the server sends the first voice file to the peer end.
[权利要求 2] 根据权利要求 1所述的语音对讲方法, 其特征在于, 所述通过卫星移 动通信网络与服务器建立连接并保持长连接的步骤之后还包括: 从所述服务器获取对端发送的第二语音文件;  [Claim 2] The voice intercom method according to claim 1, wherein the step of establishing a connection with the server and maintaining a long connection through the satellite mobile communication network further comprises: obtaining a peer transmission from the server Second voice file;
输出所述第二语音文件。  The second voice file is output.
[权利要求 3] 根据权利要求 2所述的语音对讲方法, 其特征在于, 所述从所述服务 器获取对端发送的第二语音文件的步骤包括: [Claim 3] The voice intercom method according to claim 2, wherein the step of acquiring the second voice file sent by the peer from the server comprises:
接收所述服务器发送的所述第二语音文件的下载地址;  Receiving a download address of the second voice file sent by the server;
根据所述下载地址下载所述第二语音文件。  Downloading the second voice file according to the download address.
[权利要求 4] 根据权利要求 2所述的语音对讲方法, 其特征在于, 所述输出所述第 二语音文件的步骤包括: [Claim 4] The voice intercom method according to claim 2, wherein the step of outputting the second voice file comprises:
判断所述第二语音文件是否为低速率语音编码文件;  Determining whether the second voice file is a low rate voice code file;
当为低速率语音编码文件吋, 利用低速率语音解码算法对所述第二语 音文件进行解码后予以播放。  When the speech file is a low rate speech code, the second speech file is decoded using a low rate speech decoding algorithm and played.
[权利要求 5] 根据权利要求 1所述的语音对讲方法, 其特征在于, 所述低速率语音 编码算法为自适应多速率 AMR算法、 混合激励线性预测编码 MELP算 法、 码激励线性预测编码 CELP算法、 正弦变换编码 STC算法、 吋频 域插值编码 TFI算法、 基音同步激励线性预测编码 PSELP算法、 多带 激励编码 MBE算法或波形内插编码 WI算法。  [Claim 5] The voice intercom method according to claim 1, wherein the low rate speech coding algorithm is an adaptive multi-rate AMR algorithm, a hybrid excitation linear prediction coding MELP algorithm, and a code excitation linear prediction coding CELP. Algorithm, sinusoidal transform coding STC algorithm, chirp frequency domain interpolation coding TFI algorithm, pitch synchronization excitation linear prediction coding PSELP algorithm, multi-band excitation coding MBE algorithm or waveform interpolation coding WI algorithm.
[权利要求 6] 根据权利要求 5所述的语音对讲方法, 其特征在于, 当所述低速率语 音压缩编码算法为 AMR算法吋, 所述利用低速率语音压缩编码算法 对采集的语音信息进行压缩处理并生成语音文件的步骤包括: 利用所述 AMR算法对采集的语音信息进行压缩编码, 以降低所述语 音信息的码率至预设值, 并生成 AMR格式的语音文件。 [Claim 6] The voice intercom method according to claim 5, wherein when the low rate speech compression coding algorithm is an AMR algorithm, the low rate speech compression coding algorithm is used to perform the collected speech information. The steps of compressing and generating a voice file include: The collected voice information is compression-encoded by using the AMR algorithm to reduce the code rate of the voice information to a preset value, and generate a voice file in the AMR format.
根据权利要求 6所述的语音对讲方法, 其特征在于, 所述预设值为 6.6 kb/s。 The voice intercom method according to claim 6, wherein the preset value is 6.6 kb/s.
根据权利要求 5所述的语音对讲方法, 其特征在于, 当所述低速率语 音压缩编码算法为 MELP算法吋, 所述利用低速率语音压缩编码算法 对采集的语音信息进行压缩处理并生成语音文件的步骤包括: 利用所述 MELP算法对采集的语音信息进行压缩编码, 以降低所述语 音信息的码率至 2.4kb/s, 并生成 MELP格式的语音文件。 The voice intercom method according to claim 5, wherein when the low rate speech compression coding algorithm is an MELP algorithm, the low-rate speech compression coding algorithm compresses the collected speech information and generates a speech. The step of the file includes: compressing and encoding the collected voice information by using the MELP algorithm to reduce the bit rate of the voice information to 2.4 kb/s, and generating a voice file in the MELP format.
根据权利要求 1所述的语音对讲方法, 其特征在于, 所述向所述服务 器发送所述语音文件的步骤包括: The voice intercom method according to claim 1, wherein the step of transmitting the voice file to the server comprises:
采用 TCP/IP协议分包传输所述语音文件至所述服务器。 The voice file is transmitted to the server in a packet using a TCP/IP protocol.
一种语音对讲装置, 其特征在于, 包括: A voice intercom device, comprising:
连接模块, 用于通过卫星移动通信网络与服务器保持长连接; 处理模块, 用于利用低速率语音编码算法对采集的语音信息进行压缩 处理并生成第一语音文件; a connection module, configured to maintain a long connection with the server through the satellite mobile communication network; and a processing module, configured to compress the collected voice information by using a low-rate voice coding algorithm and generate a first voice file;
发送模块, 用于向所述服务器发送所述第一语音文件, 以使所述服务 器将所述第一语音文件发送给对端。 And a sending module, configured to send the first voice file to the server, so that the server sends the first voice file to the peer end.
根据权利要求 10所述的语音对讲装置, 其特征在于, 所述装置还包括 获取模块, 用于从所述服务器获取对端发送的第二语音文件; 输出模块, 用于输出所述第二语音文件。 The voice intercom apparatus according to claim 10, wherein the apparatus further comprises: an acquisition module, configured to acquire a second voice file sent by the peer end from the server; and an output module, configured to output the second voice file Voice file.
根据权利要求 11所述的语音对讲装置, 其特征在于, 所述获取模块包 括: The voice intercom apparatus according to claim 11, wherein the acquisition module comprises:
接收单元, 用于接收所述服务器发送的所述第二语音文件的下载地址 下载单元, 用于根据所述下载地址下载所述第二语音文件。 a receiving unit, configured to receive a download address downloading unit of the second voice file sent by the server, to download the second voice file according to the download address.
根据权利要求 11所述的语音对讲装置, 其特征在于, 所述输出模块包 括: The voice intercom device according to claim 11, wherein said output module package Includes:
判断单元, 用于判断所述第二语音文件是否为低速率语音编码文件; 播放单元, 用于当为低速率语音编码文件吋, 利用低速率语音解码算 法对所述第二语音文件进行解码后予以播放。 a determining unit, configured to determine whether the second voice file is a low-rate voice encoding file, and a playing unit, configured to: after the low-rate voice encoding file, decode the second voice file by using a low-rate voice decoding algorithm Play it.
根据权利要求 10所述的语音对讲装置, 其特征在于, 所述低速率语音 编码算法为自适应多速率 AMR算法、 混合激励线性预测编码 MELP算 法、 码激励线性预测编码 CELP算法、 正弦变换编码 STC算法、 吋频 域插值编码 TFI算法、 基音同步激励线性预测编码 PSELP算法、 多带 激励编码 MBE算法或波形内插编码 WI算法。 The voice intercom device according to claim 10, wherein the low rate speech coding algorithm is an adaptive multi-rate AMR algorithm, a hybrid excitation linear predictive coding MELP algorithm, a code excited linear predictive coding CELP algorithm, and a sinusoidal transform coding. STC algorithm, 吋 frequency domain interpolation coding TFI algorithm, pitch synchronization excitation linear prediction coding PSELP algorithm, multi-band excitation coding MBE algorithm or waveform interpolation coding WI algorithm.
根据权利要求 14所述的语音对讲装置, 其特征在于, 当所述低速率语 音压缩编码算法为 AMR算法吋, 所述处理模块用于: The voice intercom apparatus according to claim 14, wherein when the low rate speech compression coding algorithm is an AMR algorithm, the processing module is configured to:
利用所述 AMR算法对采集的语音信息进行压缩编码, 以降低所述语 音信息的码率至预设值, 并生成 AMR格式的语音文件。 The collected voice information is compression-encoded by the AMR algorithm to reduce the bit rate of the voice information to a preset value, and generate a voice file in the AMR format.
根据权利要求 15所述的语音对讲装置, 其特征在于, 所述预设值为 6.The voice intercom device according to claim 15, wherein the preset value is 6.
6kb/s。 6kb/s.
根据权利要求 14所述的语音对讲装置, 其特征在于, 当所述低速率语 音压缩编码算法为 MELP算法吋, 所述处理模块用于: The voice intercom apparatus according to claim 14, wherein when the low rate voice compression coding algorithm is a MELP algorithm, the processing module is configured to:
利用所述 MELP算法对采集的语音信息进行压缩编码, 以降低所述语 音信息的码率至 2.4kb/s, 并生成 MELP格式的语音文件。 The collected voice information is compression-coded by the MELP algorithm to reduce the bit rate of the voice information to 2.4 kb/s, and generate a voice file in the MELP format.
根据权利要求 10所述的语音对讲装置, 其特征在于, 所述发送模块用 于: 采用 TCP/IP协议分包传输所述语音文件至所述服务器。 The voice intercom apparatus according to claim 10, wherein the transmitting module is configured to: packet transmit the voice file to the server by using a TCP/IP protocol.
一种移动终端, 包括存储器、 处理器和至少一个被存储在所述存储器 中并被配置为由所述处理器执行的应用程序, 其特征在于, 所述应用 程序被配置为用于执行权利要求 1所述的语音对讲方法。 A mobile terminal comprising a memory, a processor and at least one application stored in the memory and configured to be executed by the processor, wherein the application is configured to execute the claims The voice intercom method described in 1.
PCT/CN2017/109187 2017-06-12 2017-11-02 Speech talkback method and apparatus, and mobile terminal WO2018227854A1 (en)

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