TWI278219B - System for adjusting speech rate of voice over Internet protocol phone and method thereof - Google Patents

Abstract

A system for adjusting audio rate of voice over Internet protocol (VoIP) phone and a method thereof are disclosed. When the called end inputs speed parameters for adjusting the voice playing of the VoIP phone, an adjustment module can adjust the voice data in the random access memory based on the configured speed parameters, a central processor is provided for encoding/decoding, an output module outputs the adjusted voice signal according to the speed parameters, a prompt module of the calling end shows that the called end is starting the adjusted voice speed mode. Accordingly, communication is achieved with appropriate speed and clear voice while in dialog, and the calling end is also prompted.

Description

1278219 IX. Description of the Invention: [Technical Field] The present invention relates to a system for adjusting voice speed by a network telephone and a method thereof, and particularly for processing an instant voice signal, so that the user can decide to listen according to personal habits. Voice speed system and method. [Prior Art] Traditionally, the method of transmitting sound is to first establish an inherent circuit 'and then make a call in this circuit. The disadvantage is that after the transmission resource is occupied, it is necessary to wait for the - break line' to be re-routed. Used by others. But with VoIP (Voice over Intemet pr〇t〇c〇1) technology, the sound is first digitized, and then the digital sound data is cut into many small units, each small unit plus IP (Internet Protocol) After forming a packet. After the appropriate packets are transmitted to the lp-based data networks, the appropriate transmission path selection can be made according to the usage of the network; after reaching the destination, the packets are merged again and restored to the original. the sound of. With this technology, voice packets can be transmitted to the world via the Internet', but the traditional VoIP technology is quite rough, and the VoIP technology is quite rough. limit. For example, 'you can't use a regular phone for VoIP calls, but you can only use a computer as a communication device. In addition, the quality of the sound is also unstable. The total 1278219 depends on the congestion of the Internet at the time, but because of the use of ν〇π>, it can communicate with the world without paying long-distance telephone charges. More and more people are using this technology. In 1998, we introduced a switch that can integrate telecommunications networks. (There is a VoIP technology that is stepping into the integration with the traditional public switched telephone network. VGlP technology, telecommunications companies can use the Internet for long distance calls. The transmission backbone has further reduced the cost of long-distance calls. However, as the technology of the replacement is becoming more and more mature, many multinational companies have abandoned traditional long-distance calls and established a voice transmission network within the company. On the other hand, Cooperating with different regions or countries (4), the power generation policy has also created - some small telecommunications companies have emerged, they have flexible use of v〇ip technology, _ recorded ^ (four) stomach #. In the past when talking on the phone 'often because of external noise or It is the habit of calling the other party too fast (4) to hear the unclear _, or because the language is the same as the cabinet, and the appeal must be repeated by the other party to listen to the same situation. Especially through the transmission of the voice of the packet often Because of the lack of bandwidth and other reasons, there are intermittent situations. Therefore, how to adjust the speed of the dialogue reception, so that the sound is clearer and clear, also conveyed to Gu Yi, In particular, Yumi slows down the speed of receiving, or shortens the forest to the money, so that when the two sides make a call, they can rely on the individual mosquitoes to catch the degree, so that the listener can know more clearly the different words and speed.                                                 Voice signal for voice! Light, then voice output, so that the use of the receiving end = can get (four) # _ effect, and the caller can also _ call the end to adjust the voice speed notice, ton according to the appropriate adjustment .

Therefore, in order to achieve the above objective, a system for adjusting a voice speed of a network telephone according to the present invention includes at least: a setting module for receiving a parameter for setting a voice speed adjustment by a user; The voice data packet transmitted by the compression and transmitted, and the prompt signal of the voice speed adjustment are transmitted to the calling end; the temporary memory, the voice signal transmitted by the rib storage terminal, the _ 贞 residual sound speed adjustment The operation processing '-the shouting group' ribs prompts according to the prompt signal; the adjustment module is used to uncompress the voice signal line _ and decompress _, and according to the voice speed adjustment, the voiceprint of each office Integrity adjustment; and an output module for playing the adjusted voice signal. According to the purpose of the present invention, the method of the present invention comprises the following steps: after the voice 縦 Wei Wei moves, firstly, the user is adjusted to adjust the voice thief; then, the voice is adjusted. After the function is not sent, the voice parameter of the temporary storage memory towel of the receiving end or the speaking end can be adjusted according to the light parameter, and the voice signal of each voice is adjusted; finally, the adjusted voice signal is output. . 7 1278219 The features and implementations of the present invention are described in detail below with reference to the preferred embodiments. [Embodiment] The present invention will disclose a network telephone (ν〇ΙΡ, voice 〇ver Intemet)

Protocol) A system and method for adjusting voice speed. In the following detailed description of the invention, numerous specific details However, the invention may be practiced without departing from the specific details, or may be practiced by the use of alternative elements or methods. In other instances, well-known methods, procedures, components, and circuits are not described in detail to avoid unnecessarily obscuring the invention. Please refer to "FIG. 1", which is a block diagram of the system of the present invention, which includes the following components: Temporary memory element 110, which is a random access memory (ram, Random Access Memory), for example: dynamic random access memory Take memory (DRAM,

Dynamic Random Access Memory), EDO DRAM (Extend Data Out Dynamic Random Access Memory), RDRAM (Rambus DRAM), SDRAM (Synchronous Dynamic RAM), VCM SDRAM (Virtual Channel Memory SDRAM), and recently become the mainstream of the city's double data The DDR (Double Date Rate) SDRAM is used as a temporary storage of the audio stream data received by the transmission module 140. In addition to receiving the incoming voice packet from the originating end, the transmission module 140 can also receive 1278219 to receive and transmit the prompt of the voice adjustment speed set by the receiving end. When the voice data is cut into multiple packets and transmitted, the header address will be added to the network address of the receiving terminal and the related information for voice data reorganization, so that the green insurance data is correct. Therefore, an important criterion for the use of the (4) service is the signal transmission protocol gnalingpr〇t〇c〇i) to create a connection between the client's software and hardware on the network. The main function package of the call setup and control of the conversation is the address plane, address translation, connection establishment, service feature difference, call termination and call participant management.

The VoIP standard group is located in ITU_T, the Internet Engineering Task Force (IETF) and the European Telecommunications Standards Institute (ETSI). Two notable standards that can be used for positive telephone signal transmission are the H.323 series of standards and the membership agreement for positive TFs (Sessi〇n out of the clamp (10) Protocol, SIP). The agreement was originally an agreement for the development of multimedia conferences on the Internet. The two protocols H.323 and SIP represent two different solutions to the same problem. In addition, two signal transmission protocols are considered as part of the sip structure. These two protocols are: Session Description Protocol (Sessi〇n Description Protoc〇l, SDP) and Session Notification Agreement (Sessi〇n)

Announcement Protocol, SAP).

The call setup and control of VoIP are mostly based on TCP, and the transmission of audio stream is based on UDP. In order to ensure the immediacy of transmission, ietf 1278219 has added an important protocol. RSVP (Resource Reservation Protocol) ): In general, it is very difficult to reserve enough bandwidth for multimedia transmission on the Internet. The IETF defines the Resource Reservation Setup Protocol (RSVP). RSVP allows recipients to apply for a specific amount of bandwidth for data transfer, which results in a QoS (Quality of Service) guarantee. The setting module 120 is configured to receive input settings made by the user from a keyboard or other input device, for example, the user can select from the key to adjust the voice of the user to select the voice adjustment to speed up or put The multiple of slow, the start of the prompt function 'and pass the set parameters to the central processor brush for the next adjustment. The prompt module 130 is a message that when the receiving terminal starts the voice adjustment function and receives the prompt signal set by the receiving terminal, the message that can be started by the voice is displayed in the voice message, and the method can be displayed on the screen. Display, or a specific light number, or a variety of square yarns such as sound effects, etc. as a way to prompt the user to adjust the voice adjustment function. Adjusting the module 15Gxia as the voice can function _, _ received the voice call from her (10) at the receiving end, according to the parameters of the 1 tone adjustment speed set by the use surface, copy the sound signal unit The amount of data ^ unit 'or the speaker's microphone receives the user's power... (4), observe the digital voice signal and then according to the voice transmission speed parameter of 10 1278219, the multiple times to repeat the voice signal, Or, when the module compresses and encodes the voice signal, the module can add a copy of the packet to be transmitted by the person. When the receiving party receives the packet for reorganization, it can add the paste according to the number of times of copying. The purpose of sound. The central processing unit 160 is configured to encode the voice signal by using a digital signal processing (CDSP).

(voicecoding) and voice compression technology (5) (4) Qing (four) ^ voice signal coding becomes the voice of the voice, after the shrinkage, and then _ data into packets (Packetization), each packet is transmitted independently on the data network. At the receiving end, the received packet is reassembled (Paeket As_b^), the packet format (De-Paketization) is decompressed, and the digital voice signal is restored to an analog signal, so that the voice transmission function is achieved. When the user wants to speed up the playback speed of the voice, the central processor 16 〇 will receive the voice signal according to the adjustment multiple times, when the speed is 2 times the speed %, then choose to discard the two consecutive voice signals. A voiceprint signal reduces the amount of data played by voice signals by a factor of two, so the overall voice playback speed can be increased. Similarly, when the user sets the playback speed to a slower speed of 2x, it copies each voiceprint signal once and shortens the blank time between each sentence. If necessary, the overall playback time can be extended. The output module 170 herein refers to an expander on the VoIP for dialing a digit apostrophe. Please refer to "2a. Figure", which is a schematic diagram of the normal broadcast speed of the voice broadcast 11 1278219, assuming that the received voice signal contains three sentences ·, Hello, is the Smith of Xiamen,,," I would like to ask Murakami Mr. is there, every sentence, Γ 段 - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - After the voiceprint signal, the time between each sentence will be shortened: even the overall playback time of the three sentences will be longer than the normal time, because the prompting module 130 can know that the receiving terminal is starting the voice. Adjust the function, so you can know that the answer to the receiving end will be slower than the normal conversation. Similarly, when the user of the receiving user adds the speed of the _, as shown in the "2nd,", then the root 2 speeds up the speed parameter, and the scale of each sentence is shortened, maintaining the start of the car. Time, no between the Peng sentence (four) · increase. Ming Mai's "Day 3" "This is the square wire paste for voice adjustment at the receiving end of the present invention. After the voice function is activated for the receiving terminal, first, the setting module (10) of the receiving end receives the playback setting of the user's voice adjustment 'fastening or slowing down, and adjusting the speed parameter (step 3ig); then The transmission module will just send a prompt to initiate the voice adjustment function to the caller (the prompt module 130 of the chat terminal can message or prompt the light or sound to prompt the user of the caller. The transfer module U0 will Received voice packets, re-composed into a digital voice signal pre-existing (four) record (four)! Finance, tank pool (9) re-domain adjustment speed parameters, the voice signal of the temporary memory 11G voice signal into the 12 1278219 line Adjusting (step 330). For example, when the speed parameter is 2 times faster, the voice signal selected to discard one unit in two consecutive voice signals is selected, and the amount of data for playing the voice signal is reduced by - times, so the overall The voice playback speed can be accelerated. Finally, the output module 170 outputs the adjusted digital voice signal (step 340), and the user self-transmitting wheel module of the receiving end receives the voice signal. The sound of the uttered end can be heard by the output module 170. By the method disclosed in the present invention, the user can easily reduce the speed of the speaker's wire when using the call, and adjust the received voice signal. The playback speed 'and the utterer can obtain the prompt for the voice-activated function to be activated at the receiving end. Although the present invention has been disclosed above in the preferred embodiment, it is not used for the invention of the invention, any reduction (four), In the spirit and scope of the gamma, the scope of the patent protection of the present invention shall be determined by the scope of the patent application attached to the specification. [Simplified illustration] Figure 1 The system architecture diagram of the present invention; 苐2a diagram is a schematic diagram of normal playback of the present invention; 2b is a schematic diagram of slow playback of the present invention; 苐2c diagram is a schematic diagram of accelerated playback of the present invention; and FIG. 3 is a schematic diagram of the present invention. Flow chart of the method of the embodiment of the terminal. [Description of main component symbols] 110 temporary storage memory 1278219 120 setting module 130 prompting module 140 transmission module 150 adjusting module 160 central processing unit 170 output module step 310 receiving the user's voice adjustment playback setting step 320 transmitting the prompt to initiate the voice adjustment function to the utterance step 330 copying the different number of voiceprint signal unit data according to the voice adjustment speed parameter step 340 Output adjusted voice signal 14

Claims (1)

1278219 X. The scope of application for patents: 1. A system for adjusting the voice speed of a network telephone, which is applied to a voice terminal and a voice call. The edge system has: a setting module for receiving a user to set a plurality of pens a voice speed adjustment parameter; a transmission module, configured to receive a plurality of voice data packets that are compressed and encoded by the voice terminal, and reassemble into a voice signal, and transmit each voice speed adjustment parameter to the voice terminal, where The voice signal is composed of a plurality of voice signals; a temporary memory for temporarily storing the voice signal; and an adjustment module for adjusting parameters according to the voice speeds for performing the voice signals Copying and deletion of the amount of data of the unit. 2. If you apply for a patent scope! The system of claim 2, wherein the system further comprises a prompting module for prompting according to each of the voice speed adjustment parameters. 1 The system of claim 1, wherein the system further comprises: a central processing benefit compression encoding of the ribbed read voice signal, and a reorganization of each of the voice data packets. 4. The system of claim 1, wherein the system further comprises an output module for playing the copied or deleted voicing signals. The secret of the Shishen β patent ΐϋΐι item 1 is that the voice speed adjustment parameters of the towel are included in the cycle, the start of the cycle, the setting of the quick adjustment or the setting of the can and the adjustment of the number of 15 1278219. 6. A method for spoofing voice speed of a network telephone, wherein the receiving end receives a plurality of voice data packets that are compressed and encoded, and is reorganized into a voice signal to store a temporary memory, the method The method includes the following steps: receiving a user input setting a plurality of voice speed adjustment parameters; capturing the voice signal of the temporary memory and distinguishing into a plurality of voiceprint signals; According to each of the voice speed adjustment parameters, the unit data amount of each of the voiceprint signals is adjusted. 7. The method of claiming the patent is as described in the sixth aspect, wherein each of the voice speed adjustment parameters includes an adjustment function start-up, a fast-adjustment or a slow-down setting, and an adjustment multiple. 8. If you apply for the method described in the 7th handle, the parameter of the adjustment function is transmitted to the utterance end, and the prompting module of the utterance end can make a prompt. 9. If the application is as described in item (4) of the application, the party of the voice signal is the slave of the domain or the weaving, and the amount of the unit of the signal is deleted and copied. 10. The method of claim 9, wherein the unit data amount is 16