WO2009049539A1 - Procédé et matériel d'appel - Google Patents

Procédé et matériel d'appel Download PDF

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Publication number
WO2009049539A1
WO2009049539A1 PCT/CN2008/072637 CN2008072637W WO2009049539A1 WO 2009049539 A1 WO2009049539 A1 WO 2009049539A1 CN 2008072637 W CN2008072637 W CN 2008072637W WO 2009049539 A1 WO2009049539 A1 WO 2009049539A1
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WO
WIPO (PCT)
Prior art keywords
frame
rate
received
voice
speech
Prior art date
Application number
PCT/CN2008/072637
Other languages
English (en)
Chinese (zh)
Inventor
Ming Li
Jiang GUO
Guohong Li
Original Assignee
Huawei Technologies Co., Ltd.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2009049539A1 publication Critical patent/WO2009049539A1/fr

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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/04—Speech 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
    • G10L19/16—Vocoder architecture
    • G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes

Definitions

  • the present invention relates to the field of communications, and in particular, to a call method and apparatus. Background technique
  • voice communication can be implemented by Adaptive Multi-Rate (AMR) codec service.
  • AMR Adaptive Multi-Rate
  • the rate can be Adjust according to the situation.
  • the mobile phone adds a Codec Mode Indication (CMI) to the voice frame, and CMI is used to indicate the rate of the voice frame.
  • CMI Codec Mode Indication
  • CMR encoder mode request to the voice frame
  • CM is used to indicate the encoding rate of the peer.
  • CMI and CM can be transmitted in alternate intervals, such as CMI in the first frame, CMR in the second frame, CMI in the third frame, CMR in the fourth frame, and CMI in the second frame.
  • the default is equal to the CMI of the first frame
  • the CMI of the fourth frame is equal to the CMI of the third frame by default, that is, the rate at which the CMR is transmitted is the same as the frame rate of the CMI transmitted in the previous frame.
  • the network side After the Internet Protocol (IP), the network side transmits the received voice IP packet to the mobile phone, the CMI and the CM are encapsulated in one voice IP packet, and the voice frame between the network side and the mobile phone is adopted.
  • the CMI and the CMR are transmitted in an alternate manner. If the CMI of the voice IP packet received by the network side is different from the CMI of the voice IP packet of the previous frame when the network side sends the CMR to the mobile phone, the network side cannot send the voice frame to Mobile phones, which affect the quality of voice communication, and even interrupt voice communication.
  • An object of the embodiments of the present invention is to provide a method and apparatus for calling, which implements reliable transmission of voice and improves the quality of voice communication.
  • the embodiment of the invention provides a method for calling, the method comprising:
  • the received even speech frame stores the odd duration of the speech frame.
  • the embodiment of the invention further provides a device for talking, the device comprising:
  • a receiving unit configured to receive a voice frame
  • a determining unit configured to determine whether a rate of the even-numbered voice frames in the voice frame received by the receiving unit is consistent with a rate of the previously received voice frame
  • a storage unit configured to: when the determining unit determines that the result is inconsistent, store the received even-numbered voice frames for an odd duration of the voice frame.
  • the network side cannot transmit the voice frame, and the call method and apparatus according to the embodiment of the present invention receive an even-numbered voice frame.
  • the even-numbered speech frame received by the network side stores the odd duration of the speech frame, so that the even-numbered speech frame is stored and converted into an odd-numbered speech frame, and the network side can directly The odd voice frames are sent to the terminal, thereby realizing reliable transmission of voice and improving the quality of voice communication.
  • Embodiment 1 is a schematic flow chart of Embodiment 1 of a call method according to the present invention.
  • FIG. 2 is a schematic structural diagram of an embodiment of a device for talking in the present invention. detailed description
  • Embodiment 1 discloses a call method. Referring to FIG. 1, the method includes:
  • the frame number and rate of the received speech frame may be recorded. If the frame number of the speech frame is an even number, compare the speech frame. Whether the rate is consistent with the rate of the previous received speech frame.
  • the rate of receiving the even-numbered speech frame is determined. The rate at which the previous received voice frame was inconsistent;
  • the odd-numbered voice frame sent by the uplink is inconsistent with the previous transmitted voice frame rate, and the frame difference between the uplink-transmitted voice frame number and the corresponding downlink-received voice frame number is an even number, it is determined that the even-numbered voice frame is received.
  • the rate does not match the rate at which the previous received speech frame was received.
  • the method for determining the frame difference between the voice frame number sent by the uplink and the voice frame number received by the downlink may be:
  • Detecting the condition of the channel that is, detecting the delay of the channel
  • the frame difference between the voice frame number sent by the uplink and the voice frame number received by the downlink is determined according to the delay condition of the channel.
  • the received even speech frame stores the odd duration of the speech frame.
  • the network side cannot transmit the voice frame, which may cause the voice call to be interrupted, and the call method and apparatus disclosed in the embodiment of the present invention are used.
  • the network side detects whether the rate of receiving the even-numbered speech frame is consistent with the rate of the previous received speech frame, and detects the rate of the received speech frame. If not, the received even-numbered speech frame stores the odd duration of the speech frame. In this way, the even-numbered speech frame is stored and converted into an odd-numbered speech frame, and the network side can directly transmit the odd-numbered speech frame to the terminal, thereby realizing reliable transmission of the voice and improving the quality of the voice communication.
  • the even-numbered speech frame stores a frame length of one speech frame, and the frame length of one speech frame is twenty milliseconds, and the speech frame is delayed by twenty milliseconds, and the user who uses the voice call can hardly feel the delay. Therefore, there is no influence on the call quality, and reliable transmission of voice is realized.
  • the call method disclosed in the first embodiment can be applied to the variable rate voice communication.
  • the call method disclosed in the first embodiment is applied to the adaptive multi-rate codec of the global mobile communication system.
  • the odd length of the stored voice frame is the frame length of one voice frame.
  • the frame length of a voice frame is twenty milliseconds.
  • the call method in this embodiment includes: 1.
  • the base station receives the voice frame of the mobile phone uplink or system downlink, and reserves a buffer of twenty milliseconds in the place where the base station receives the downlink voice frame of the system, that is, reserves a frame of the voice frame; when the voice channel is established, the base station Send CMR to the system uplink, and send the rule to two frames as an example.
  • the first, third, fifth... frame time corresponds to the CMR frame time received from the mobile phone uplink, marked as odd (0) frame time
  • the sixth frame time corresponds to the CMI frame time received from the mobile phone uplink, and is marked as an even (E) frame time.
  • Two rates for example, 7.4K and 6.7K, are selected, and the transmission is performed alternately, that is, the first and second frame rates are 7.4k, the third and fourth frame rate is 6.7k, and the fifth and sixth frame rates are 7.4k.
  • the system After receiving the CMR request from the base station, the system will change the coding rate to the base station, that is, change the system direction. CMI of the downlink voice frame of the base station.
  • the base station After the base station sends a regularly changing CMR to the system, it starts monitoring the downlink received from the system.
  • the base station determines whether the voice frame CMI currently received by the system is consistent with the received voice frame CMI of the previous frame, if the frame number of the currently received voice frame is an odd number, and the currently received voice frame CMI and the previous frame If the CMIs of the received voice frames are inconsistent, the frame difference caused by the delay of the voice channel is even, and if it is consistent, the frame difference is an odd number.
  • the frame difference caused by the voice channel delay is an odd number, and vice versa. Consistent, the frame difference is even.
  • the determination is received.
  • the rate of the even speech frame is inconsistent with the rate of the previous received speech frame.
  • an even-numbered speech frame whose rate changes can be converted into an odd-numbered speech frame to ensure normal operation of the voice communication.
  • the device includes:
  • the receiving unit 201 is configured to receive a voice frame.
  • the determining unit 202 is configured to determine whether the rate of the even-numbered voice frames in the voice frame received by the receiving unit is consistent with the rate of the previously received voice frame;
  • the storage unit 203 is configured to: when the determining unit 202 determines that the result is inconsistent, store the received even-numbered speech frames for an odd duration of the speech frame.
  • the device of the call may further include:
  • a detecting unit configured to detect a channel condition
  • a frame difference unit configured to determine, according to the detection result of the detecting unit, a frame difference between the voice frame number sent by the uplink and the voice frame number corresponding to the downlink.
  • the determining unit may include:
  • a first module configured to: when an even-numbered voice frame sent in the uplink is inconsistent with a previously transmitted voice frame rate, and the frame difference determined by the frame difference unit is an odd number, determine a rate at which the even-numbered voice frame is received and the previous received The rate of the voice frames is inconsistent; or,
  • a second module configured to: when an odd voice frame sent in the uplink is inconsistent with a previously transmitted voice frame rate, and the frame difference determined by the frame difference unit is an even number, determine a rate at which an even voice frame is received and a previous reception The rate to the speech frame is inconsistent.
  • the device of the call can be applied in variable-rate voice communication, set on the network side, can be set in the base station, or can be set in the gateway, or can be used as an independent device to cooperate with the base station or the gateway to implement variable-rate voice. Reliable operation of communications.

Abstract

L'invention porte sur un procédé et un matériel d'appel. Le procédé comprend : la réception d'une trame vocale ; le stockage de la trame vocale paire reçue dans une période impaire de trame vocale lorsque le débit de la trame vocale paire reçue n'est pas identique au débit d'une trame vocale reçue précédente. Par utilisation du procédé et du matériel d'appel décrits dans le mode de réalisation de la présente invention, en déterminant si le débit de la trame vocale paire reçue est identique au débit d'une trame vocale reçue précédente, le côté réseau détecte le débit d'une trame vocale reçue, si les résultats détectés ne sont pas identiques, alors il stocke la trame vocale paire reçue dans une période impaire de trame vocale, ainsi la trame vocale paire est transformée en une trame vocale impaire, et le côté réseau peut envoyer la trame vocale à un terminal directement.
PCT/CN2008/072637 2007-10-12 2008-10-10 Procédé et matériel d'appel WO2009049539A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN200710163580.6 2007-10-12
CN2007101635806A CN101409597B (zh) 2007-10-12 2007-10-12 一种通话方法和装置

Publications (1)

Publication Number Publication Date
WO2009049539A1 true WO2009049539A1 (fr) 2009-04-23

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PCT/CN2008/072637 WO2009049539A1 (fr) 2007-10-12 2008-10-10 Procédé et matériel d'appel

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CN (1) CN101409597B (fr)
WO (1) WO2009049539A1 (fr)

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102790997B (zh) * 2011-05-19 2017-05-10 中兴通讯股份有限公司 一种自适应多速率amr语音数据的传输方法及装置

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1549470A (zh) * 2003-05-13 2004-11-24 西门子（中国）有限公司 Tdma移动通信系统中扩大测量窗口的方法
JP2007121330A (ja) * 2005-10-24 2007-05-17 Dainippon Printing Co Ltd 音響信号に対する情報の埋め込み装置
CN1991977A (zh) * 2005-12-29 2007-07-04 Ut斯达康通讯有限公司 用于无线通信的语音编码速率确定方法

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1549470A (zh) * 2003-05-13 2004-11-24 西门子（中国）有限公司 Tdma移动通信系统中扩大测量窗口的方法
JP2007121330A (ja) * 2005-10-24 2007-05-17 Dainippon Printing Co Ltd 音響信号に対する情報の埋め込み装置
CN1991977A (zh) * 2005-12-29 2007-07-04 Ut斯达康通讯有限公司 用于无线通信的语音编码速率确定方法

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CN101409597B (zh) 2011-04-13
CN101409597A (zh) 2009-04-15

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